IEEE Std 269-2001 Draft Standard Methods for Measuring

IEEE Std 269-2001 Draft Standard Methods for Measuring
For IEEE/STIT work only. Do not submit to any other standards organization.
IEEE P269/D9 Jan. 2002
IEEE
Std 269-2001
(Revision of IEEE Std 269-1992)
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).
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 of from 100 to
8,500Hz.
Keywords: analog telephones, digital telephones, headsets, handsets, electroacoustic measurements on
telephones, telephony, voice transmission performance
Copyright  2000 by the
Institute of Electrical and Electronics Engineers, Inc.
345 East 47th Street, New York, NY 10017-2394, USA
All rights reserved.
This is an unapproved draft of a proposed IEEE Standard, subject to change. Permission is hereby granted for IEEE
Standards Committee participants to reproduce this document for purposes of IEEE standardization activities. If this
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document for these or other uses must contact the IEEE Standards Department for the appropriate license. Use of
information contained in the unapproved draft is at your own risk.
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IEEE P269/D9 Jan. 2002
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Introduction
(This introduction is not a part of IEEE Std 269-2000, 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
Cliff Chamney
Paul Coverdale
Fred Dekalb
Dan Foley
James Gurnavage
Roger Gutzwiller
Roger Hunt
Frederick M. Kruger
Ron Magnuson
Christopher Struck
Stephen Whitesell
Allen Woo
Robert Young
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Contents
1
2
3
4
5
6
7
Overview........................................................................................................................................................ 9
1.1 Scope ....................................................................................................................................................... 9
1.2 Purpose .................................................................................................................................................... 9
1.3 Contents of standard................................................................................................................................. 9
References.................................................................................................................................................... 11
Definitions.................................................................................................................................................... 12
Abbreviations, Acronyms and Symbols......................................................................................................... 15
4.1 Abbreviations and acronyms................................................................................................................... 15
4.2 Symbols ................................................................................................................................................. 15
Test Equipment and Setup ............................................................................................................................ 17
5.1 Ear simulators ........................................................................................................................................ 17
5.2 Mouth simulators.................................................................................................................................... 18
5.3 Test Fixtures........................................................................................................................................... 18
5.3.1
Test fixtures .................................................................................................................................. 18
5.3.2
Handset positioning....................................................................................................................... 18
5.3.3
Headset positioning ....................................................................................................................... 19
5.4 Measurement microphones ..................................................................................................................... 21
5.5 Test environment.................................................................................................................................... 21
5.5.1
Background noise level.................................................................................................................. 22
5.5.2
Free field conditions...................................................................................................................... 22
5.5.3
Diffuse field conditions ................................................................................................................. 22
5.5.4
Reference corner ........................................................................................................................... 23
5.6 Acoustic impairments............................................................................................................................. 24
5.6.1
Hoth room noise............................................................................................................................ 24
Test Calibration ............................................................................................................................................ 25
6.1 Measurement bandwidth and resolution .................................................................................................. 25
6.2 Send....................................................................................................................................................... 25
6.2.1
Acoustic test spectrum................................................................................................................... 25
6.2.2
Acoustic test level ......................................................................................................................... 25
6.2.3
Mouth simulator calibration procedure........................................................................................... 25
6.3 Receive and echo frequency response ..................................................................................................... 26
6.3.1
Electrical test spectrum.................................................................................................................. 26
6.3.2
Electrical test level ........................................................................................................................ 26
Test Procedure for Analog Sets..................................................................................................................... 28
7.1 General .................................................................................................................................................. 28
7.1.1
Choice of test signals and levels..................................................................................................... 28
7.1.2
Measurement bandwidth and resolution ......................................................................................... 28
7.1.3
Choice of ear and mouth simulators and test position ..................................................................... 29
7.1.4
Tone control setting....................................................................................................................... 29
7.1.5
Reference receive volume control setting....................................................................................... 29
7.1.6
Reference send gain control setting................................................................................................ 29
7.2 Analog DC Feed circuits......................................................................................................................... 29
7.3 Analog telephone network impairments .................................................................................................. 31
7.3.1
Loop current.................................................................................................................................. 31
7.3.2
Network noise ............................................................................................................................... 31
7.3.3
Termination impedance ................................................................................................................. 31
7.3.4
Test loops...................................................................................................................................... 32
7.3.5
Parallel sets ................................................................................................................................... 32
7.3.6
Cordless range............................................................................................................................... 32
7.4 Receive .................................................................................................................................................. 32
7.4.1
Receive frequency response........................................................................................................... 32
7.4.2
Receive broad-band noise .............................................................................................................. 32
7.4.3
Receive narrow-band noise............................................................................................................ 33
7.4.4
Receive linearity............................................................................................................................ 33
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7.4.5
Receive distortion.......................................................................................................................... 33
7.4.6
Receive mute Needs Re-working ................................................................................................. 33
7.5 Send....................................................................................................................................................... 34
7.5.1
Send frequency response................................................................................................................ 34
7.5.2
Send broad-band noise................................................................................................................... 34
7.5.3
Send narrow-band noise................................................................................................................. 35
7.5.4
Send linearity ................................................................................................................................ 35
7.5.5
Send distortion .............................................................................................................................. 35
7.5.6
Send mute needs work.................................................................................................................. 35
7.5.7
Send frequency response in a diffuse field...................................................................................... 36
7.5.8
Send signal-to-noise ratio .............................................................................................................. 36
7.6 Sidetone ................................................................................................................................................. 37
7.6.1
Talker sidetone frequency response................................................................................................ 37
7.6.2
Listener sidetone frequency response ............................................................................................. 37
7.6.3
Alternate method for listener sidetone............................................................................................ 38
7.6.4
Sidetone linearity........................................................................................................................... 38
7.6.5
Sidetone distortion......................................................................................................................... 38
7.6.6
Sidetone delay............................................................................................................................... 39
7.6.7
Sidetone echo response.................................................................................................................. 39
7.7 Overall ................................................................................................................................................... 39
7.7.1
Overall frequency response............................................................................................................ 39
7.7.2
Overall linearity ............................................................................................................................ 39
7.7.3
Overall distortion........................................................................................................................... 40
7.8 Telephone set impedance........................................................................................................................ 40
7.8.1
AC impedance............................................................................................................................... 40
7.8.2
Return loss .................................................................................................................................... 40
7.9 Stability test ........................................................................................................................................... 40
7.10
Maximum acoustic output .................................................................................................................. 40
7.10.1 Maximum acoustic pressure (long duration).................................................................................. 41
7.10.2 Peak acoustic pressure (short duration) .......................................................................................... 41
8 Test Procedures for Digital and 4-wire Systems............................................................................................. 43
8.1 General .................................................................................................................................................. 43
8.1.1
Choice of test signals and levels..................................................................................................... 43
8.1.2
Measurement bandwidth and resolution ......................................................................................... 43
8.1.3
Choice of ear and mouth simulators and test position ..................................................................... 44
8.1.4
Tone control setting....................................................................................................................... 44
8.1.5
Reference receive volume control .................................................................................................. 44
8.1.6
Reference send gain control setting................................................................................................ 44
8.2 Digital test circuits.................................................................................................................................. 44
8.2.1
Digital telephone interface ............................................................................................................. 44
8.2.2
Reference codec ............................................................................................................................ 46
8.2.3
Wideband reference codec ............................................................................................................. 46
8.3 Digital telephone network impairments................................................................................................... 47
8.3.1
Network Delay .............................................................................................................................. 47
8.3.2
Network Packet Loss ..................................................................................................................... 47
8.3.3
Echo Canceller .............................................................................................................................. 47
8.3.4
Discontinuous Speech Transmission .............................................................................................. 47
8.4 Receive .................................................................................................................................................. 48
8.4.1
Receive frequency response........................................................................................................... 48
8.4.2
Receive broad-band noise .............................................................................................................. 48
8.4.3
Receive narrow-band noise............................................................................................................ 48
8.4.4
Receive linearity............................................................................................................................ 49
8.4.5
Receive distortion.......................................................................................................................... 49
8.4.6
Receive delay................................................................................................................................ 49
8.4.7
Receive out-of-band signals........................................................................................................... 49
8.5 Send....................................................................................................................................................... 50
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8.5.1
Send frequency response................................................................................................................ 50
8.5.2
Send broad-band noise................................................................................................................... 50
8.5.3
Send narrow-band noise................................................................................................................. 51
8.5.4
Send linearity ................................................................................................................................ 51
8.5.5
Send distortion .............................................................................................................................. 51
8.5.6
Send delay..................................................................................................................................... 51
8.5.7
Send out-of-band signals ............................................................................................................... 52
8.5.8
Send frequency response in a diffuse field...................................................................................... 52
8.5.9
Send signal-to-noise ratio .............................................................................................................. 52
8.6 Sidetone ................................................................................................................................................. 53
8.6.1
Talker sidetone frequency response................................................................................................ 53
8.6.2
Listener sidetone frequency response ............................................................................................. 53
8.6.3
Alternate method for listener sidetone............................................................................................ 54
8.6.4
Sidetone linearity........................................................................................................................... 54
8.6.5
Sidetone distortion......................................................................................................................... 54
8.6.6
Sidetone delay............................................................................................................................... 55
8.6.7
Sidetone echo response.................................................................................................................. 55
8.7 Overall ................................................................................................................................................... 55
8.7.1
Overall frequency response............................................................................................................ 55
8.7.2
Overall linearity ............................................................................................................................ 56
8.7.3
Overall distortion........................................................................................................................... 56
8.8 Echo frequency response ........................................................................................................................ 56
8.9 Stability loss........................................................................................................................................... 57
8.10
Convergence time .............................................................................................................................. 57
8.11
Discontinuous speech transmission..................................................................................................... 57
8.11.1 General ......................................................................................................................................... 58
8.11.2 Measurement method .................................................................................................................... 58
8.12
Maximum acoustic output .................................................................................................................. 58
8.12.1 Maximum acoustic pressure (long duration).................................................................................. 58
8.12.2 Peak acoustic pressure ( short duration).......................................................................................... 59
9 Test Procedures for Analog 4-wire Headsets and Handsets ............................................................................ 61
9.1 General .................................................................................................................................................. 61
9.1.1
Choice of test signals and levels..................................................................................................... 61
9.1.2
Measurement bandwidth and resolution ......................................................................................... 61
9.1.3
Choice of ear and mouth simulators and test position ..................................................................... 62
9.1.4
Tone control setting....................................................................................................................... 62
9.1.5
Default receive volume control and send gain adjustment............................................................... 62
9.2 Headset and handset test circuits............................................................................................................. 62
9.3 Receive .................................................................................................................................................. 64
9.3.1
General ......................................................................................................................................... 64
9.3.2
Receive volume control adjustment................................................................................................ 64
9.3.3
Receive frequency response........................................................................................................... 64
9.3.4
Receive broad-band noise .............................................................................................................. 65
9.3.5
Receive narrow-band noise............................................................................................................ 65
9.3.6
Receive linearity............................................................................................................................ 65
9.3.7
Receive distortion.......................................................................................................................... 66
9.3.8
AC impedance............................................................................................................................... 66
9.3.9
DC resistance ................................................................................................................................ 66
9.4 Send....................................................................................................................................................... 66
9.4.1
Send gain control adjustment ......................................................................................................... 66
9.4.2
Send frequency response................................................................................................................ 66
9.4.3
Send overall noise ......................................................................................................................... 67
9.4.4
Send narrow-band noise................................................................................................................. 67
9.4.5
Send linearity ................................................................................................................................ 68
9.4.6
Send distortion .............................................................................................................................. 68
9.4.7
Send frequency response in a diffuse field...................................................................................... 68
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9.4.8
Send signal-to-noise ratio .............................................................................................................. 68
9.4.9
AC impedance............................................................................................................................... 69
9.4.10 DC resistance ................................................................................................................................ 69
9.5 Echo frequency response ........................................................................................................................ 69
9.6 Maximum acoustic output....................................................................................................................... 70
9.6.1
Maximum acoustic pressure (long duration)................................................................................... 70
9.6.2
Peak acoustic pressure (short duration) .......................................................................................... 70
Annex A (normative) Ear Simulators with Flexible Pinnas and Positioning Devices........................................... 71
A.1
General characteristics of the ear simulators ....................................................................................... 71
A.2
Differences between the two ear simulators ........................................................................................ 71
A.3
Handset Positioning devices ............................................................................................................... 72
Annex B (normative) Alternative Ear Simulators, Mouth Simulator and Test Fixture......................................... 73
B.1
Alternative Ear Simulators ................................................................................................................. 73
B.2
Alternative Mouth Simulator .............................................................................................................. 75
B.3
Alternative Test Fixture...................................................................................................................... 75
Annex C (normative) DRP TO ERP Corrections ............................................................................................... 76
Annex D (normative) Conditioning for Carbon Transmitters.............................................................................. 79
Annex E (normative) Hoth Room Noise............................................................................................................ 80
Annex F (normative) Test Signals.................................................................................................................... 83
F.1 General .................................................................................................................................................. 83
F.2 Classifications ........................................................................................................................................ 83
F.3 Modulation types.................................................................................................................................... 83
F.3.1
Square wave modulation................................................................................................................ 83
F.3.2
Sine wave modulation ................................................................................................................... 84
F.3.3
Pseudo-random modulation ........................................................................................................... 84
F.4 Deterministic signals .............................................................................................................................. 84
F.4.1
Sine wave...................................................................................................................................... 84
F.4.2
Pseudo-random.............................................................................................................................. 84
F.5 Random signals ...................................................................................................................................... 84
F.5.1
White noise ................................................................................................................................... 84
F.5.2
Pink noise ..................................................................................................................................... 85
F.6 Speech-like signals................................................................................................................................. 85
F.6.1
Simulated speech........................................................................................................................... 85
F.6.2
Synthesized speech........................................................................................................................ 85
F.6.3
Real speech ................................................................................................................................... 85
F.7 Compound signals .................................................................................................................................. 86
F.7.1
Sequential presentation .................................................................................................................. 86
F.7.2
Simultaneous presentation ............................................................................................................. 86
F.8 Test signal bandwidth ............................................................................................................................. 87
F.9 Signal parameter summary...................................................................................................................... 88
F.10
Test signals published on CD-ROM ................................................................................................... 88
F.11
Signal and test method comparative summary..................................................................................... 89
Annex G (normative) Analysis Methods........................................................................................................... 90
G.1
General.............................................................................................................................................. 90
G.2
Fast Fourier transform (FFT) and cross spectrum analysis................................................................... 90
G.2.1
Dual-channel FFT ......................................................................................................................... 90
G.2.2
Single-channel FFT ....................................................................................................................... 90
G.2.3
Maximum length sequence (MLS) analysis.................................................................................... 91
G.3
Real-time filter analysis (RTA) .......................................................................................................... 91
G.3.1
Dual-channel real-time filter analysis............................................................................................. 91
G.3.2
Single-channel real-time filter analysis........................................................................................... 91
G.4
Sine-based analysis ............................................................................................................................ 91
G.4.1
Discrete tone (stepped sine) ........................................................................................................... 91
G.4.2
Swept sine..................................................................................................................................... 92
G.4.3
Time delay spectrometry (TDS)..................................................................................................... 92
G.5
Simulated free field techniques........................................................................................................... 92
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G.6
Measurement bandwidth and measurement resolution......................................................................... 92
Annex H (normative) Loudness Rating Calculations.......................................................................................... 97
Annex I (normative) Linearity ......................................................................................................................... 99
Annex J (normative) Distortion Work: sines vs narrowband random; default method? ................................... 103
J.1 Total harmonic distortion...................................................................................................................... 103
J.1.1
Test signal................................................................................................................................... 103
J.1.2
Suitability test ............................................................................................................................. 103
J.2 Sinusoidal Methods .............................................................................................................................. 104
J.2.1
Total harmonic distortion (THD) ................................................................................................. 104
J.2.2
Total Harmonic Distortion (THD) and noise ................................................................................ 104
J.2.3
Difference-frequency distortion (DF Distortion)........................................................................... 105
J.3 Alternate stimulus signals..................................................................................................................... 105
J.4 Recommended method: signal-to-distortion-and noise ratio (SDN)....................................................... 106
J.5 Continuous spectrum distortion............................................................................................................. 107
Annex K (normative) Send Signal-to-Noise Ratio............................................................................................ 108
K.1
Send signal-to-noise ratio................................................................................................................. 108
K.2
Weighted send signal-to-noise ratio.................................................................................................. 108
Annex L (normative) Delay ............................................................................................................................ 110
L.1 General ................................................................................................................................................ 110
L.2 Captured pulse method ......................................................................................................................... 110
L.3 Two-channel analyzer methods............................................................................................................. 110
L.3.1
Impulse response method............................................................................................................. 110
L.3.2
Cross-correlation method............................................................................................................. 110
L.4 Time Delay Spectrometry Method ........................................................................................................ 110
L.5 MLS Method ........................................................................................................................................ 110
Annex M (normative) Sidetone Echo ............................................................................................................... 111
Annex N (informative) Maximum Acoustic Pressure Limits............................................................................ 112
N.1
ABSTRACT .................................................................................................................................... 112
N.2
INTRODUCTION ........................................................................................................................... 112
N.3
PROPOSAL..................................................................................................................................... 114
Annex O (normative) Simulated Speech Generator (SSG) ............................................................................... 117
Annex P (normative) ITU-T Recommendation P.50 Noise Bursts Over TDS Sweep ........................................ 119
Annex Q (informative) Use of the Free Field as the Telephonometric Reference Point .................................... 120
Annex R (informative) Useful Conversion Procedures.................................................................................... 121
R.1
Conversions for dBV to dBm for 600 and 900Ω ............................................................................... 121
R.2
Conversions for dBmp to dBrnC for electrical noise measurements................................................... 122
R.3
Loudness rating conversions............................................................................................................. 122
R.4
Acoustic sound pressure conventions................................................................................................ 122
Annex S (informative) Loudness Balance Subjective Test Procedure.............................................................. 123
S.1 Introduction.......................................................................................................................................... 123
S.2 Loudness balance test procedure........................................................................................................... 123
S.3 Example test circuit .............................................................................................................................. 124
S.4 Estimate of test headset receive characteristics ...................................................................................... 125
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Draft Standard Methods for Measuring
Transmission Performance of Analog and
Digital Telephone Sets, Handsets and Headsets
1
Overview
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 they generally agree with the desirable
performance characteristics of subjective testing.
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 in
conditions encountered during normal operation.
1.1
Scope
This standard provides the techniques for objective measurement of electroacoustic characteristics of analog and
digital telephones, handsets and headsets.
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.
1.2
Purpose
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.
1.3
Contents of standard
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.
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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, 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.
Two types of annexes are placed after the body of the standard either for convenience, or to create a hierarchical
distinction. 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.
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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, American National Standard Acoustical Terminology (Including Mechanical Shock and
Vibration).
ANSI S1.4-1983, American National Standard Specification for Sound Level Meters.
ANSI S1.12-1967 (Reaff. 1986), American National Standard Specifications for Laboratory Standard Microphones.
No reference found, plus it’s covered in P.57.
[5] CCITT Blue Book, Vol. III.1 (Recommendations G.101–G.181), General Characteristics of International
Telephone Connections and Circuits.
[6] CCITT Blue Book, Vol. III.3 (Recommendations G.601–G.654), Transmission Media Characteristics.
[7] CCITT Blue Book, Vol. IV.4 (Series O Recommendations), Specifications for Measuring Equipment.
[8] CCITT Blue Book, Vol. V (Series P Recommendations), Telephone Transmission Quality.
Obsolete or duplicated by newer ITU references? Someone please CHECK?
ITU-T P.57 is the better reference.
IEEE Std 100-1992, The New IEEE Standard Dictionary of Electrical and Electronics Terms.
IEEE Std 661-1979 (Reaff. 1992), IEEE Standard Method for Determining Objective Loudness Ratings of
Telephone Connections.
IEEE Std 743-1984, IEEE Standard Methods and Equipment for Measuring the Transmission Characteristics of
Analog Voice Frequency Circuits (ANSI).
ITU-T Recommendation P.50 (1999), Artificial Voices
ITU-T Recommendation P.50, Appendix 1: Test signals (1998)
ITU-T Recommendation P.51 (1996), Artificial Mouths
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
Copyright © 2002 IEEE. All rights reserved.
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IEEE P269/D9 Jan. 2002
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 Std 100-1992
[10] and ANSI S1.1-1960 [1].
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 where 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 mouth reference point (MRP), 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 2´10-5 Pa, A-weighted (3.1).
3.9
dBm. Power level in decibels, relative to a power of 1 mw (milliwatt).
3.10
dBPa. Sound pressure level in decibels, relative to a sound pressure of 1 Pa (Pascal).
3.11
digital telephone set. A telephone set where the two-way voice communication interface to the network is
in a digital format.
3.12
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.
3.13
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.
3.14
four-wire. A transmission method, circuit or system which provides separate paths (two pairs) for signals
in the send and receive directions.
3.15
frequency response.
function of frequency.
3.16
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).
Electrical, acoustic, or electroacoustic sensitivity (output/input), or gain, as a
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3.17
ideal codec. A codec that has theoretically optimum characteristics.
3.18
listener sidetone. The signal present at the receiver due to sound in the environment where the telephone is
used.
3.19
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].
3.20
loudness rating guard-ring position-H (LRGP-H). The test position a handset assumes when it is placed
on an artificial test head as described in ITU-T Recommendation P.64, Annex E.
3.21
microphone. An electroacoustic transducer that converts sound to an electrical signal.
3.22
mouth reference point (MRP). A point on the axis of the mouth simulator, 25 mm in front of the center of
the external plane.
3.23
overall. The direction of speech transmission from the mouth of one person to the ear of another person..
Also called end-to-end.
3.24
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).
3.25
receive.The direction of speech transmission from network to ear of the telephone user.
3.26
receiver. An electroacoustic transducer that converts an electrical signal to sound and delivers it directly to
the ear, sealed or unsealed.
3.27
reference codec. A codec that approaches the performance of an ideal codec and has superior, well-defined
characteristics used for testing digital telephone sets.
3.28
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).
3.29
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.
3.30
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.
3.31
send. The direction of speech transmission from mouth to network.
3.32
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).
3.33
sidetone. The direction of speech transmission from microphone to receiver of the handset or headset..
There are two types of sidetone to be considered: listener sidetone and talker sidetone.
.
3.34
3.35
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).
soft HATS pinna. An anatomically-shaped pinna which is structurally identical to the pinna described in
ITU-T Recommendation P.57 Type 3.3, except it has a shore-A hardness of 4, +/-1. (The Type 3.3 pinna
has a shore-A hardness of 25, +-3.)
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3.36
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/m^2. An A-weighted (see ANSI S1.4-1983 (R 1997) ) sound pressure level in dB (dBSPL, A-weighted)
is often abbreviated as dBA or dB(A).
3.37
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.
3.38
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% or the center frequency ,respectively). Bands may also be of fixed width, regardless of
center frequency (e.g. 100Hz). Instead of bands, a spectrum may also be expressed as spectrum density,
which is equivalent to 1Hz bands.
3.39
talker sidetone. The direction of speech transmission from mouth to ear of the telephone user.
3.40
telephone set. A device that, when connected to a telephone network, allows two-way voice
communication.
3.41
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.
3.42
two-wire transmission. A transmission method, circuit, or system which provides common paths (one
pair) for signals in the send and receive directions
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4
Abbreviations, Acronyms and Symbols
4.1
AGC
DFTP
DRP
DSP
DTX
ERP
FFT
HATS
LRGP-H
LRPG
MRP
OLR
RETP
RLR
SETP
SFI
SLR
SNR
SNDR
STMR
TDS
THD
VAD
4.2
IEEE P269/D9 Jan. 2002
Abbreviations and acronyms
automatic gain control
diffuse field test point
drum reference point
digital signal processor
discontinuous speech transmission
ear reference point
fast Fourier transform
head and torso simulator
loudness rating guardring position-hats
loudness rating guardring position
mouth reference point
overall loudness rating
receive electrical test point
receive loudness rating
send electrical test point
single frequency interference
send loudness rating
signal-to-noise ratio
signal-to-noise and distortion ratio
sidetone masking rating
time delay spectrometry
total harmonic distortion
voice activity detector
Symbols
The letter “G” is used for spectra. This corresponds to common usage, especially in two-channel FFT analysis
literature. The analysis bandwidth shall be specified:
GDFTP(f) = spectrum at diffuse field test point (in dBPa)
GERP(f) = spectrum at Ear Reference Point (in dBPa)
GMRP(f) = spectrum at Mouth Reference Point (in dBPa)
G(MRP)(ERP)(f) = cross-spectrum between MRP and ERP (in dBPa/Pa)
G(MRP)(SETP)(f) = cross-spectrum between MRP and SETP (in dBV/Pa)
G(RETP )(f) = spectrum at Receive Electrical Test Point (in dBV)
G(RETP)(ERP)(f) = cross-spectrum between RETP and ERP (in dBPa/V)
G(RETP)(SETP)(f) = cross-spectrum RETP and SETP (in dBV/V)
GSETP(f) = spectrum at Send Electrical Test Point (in dBV)
GSETP(S+N)(f) is the RMS power spectrum at SETP with both the mouth simulator and noise sources active
GSETP(N)(f) is the RMS power spectrum at SETP with only the noise source active.
The letter “H” is used for frequency response:
H(f) = frequency response (in dB)
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) .
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The letter “L” is used for levels measured over a wide band, with the bandwidth to be specified. This corresponds to
common usage in sound level measurements, as specified in ANSI S1.1-1994:
LERP = level at Ear Reference Point (in dBPa)
LMRP = level at Mouth Reference Point (in dBPa)
LRETP = level at Receive Electrical Test Point (in dBV or dBmp)
LROOM = level of background noise in measurement environment (in dBSPL)
LSETP = level at Send Electrical Test Point (in dBV or dBmp)
The letter “S” is used for specially calculated sensitivities:
SDE = correction for HATS Drum Reference Point to Ear Reference Point
The letter “T” is used for time measurements:
T = length of time window in simulated free-field techniques (in sec)
TC = convergence time of acoustic echo cancellers (AEC) algorithm (in sec)
“TCL” is used for Terminal Coupling Loss:
TCL = terminal coupling loss (in dB)
TCLT = temporally weighted terminal coupling loss (in dB)
TCLW = frequency weighted terminal coupling loss (in dB)
TCLWST = frequency weighted terminal coupling loss during single talk (in dB)
Other symbols:
SendSNR = send signal-to-noise ratio
SendSNRW = weighted send signal-to-noise ratio
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IEEE P269/D9 Jan. 2002
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 4-wire devices such as headsets and handsets, 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.
5.1
Ear simulators
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.
An ear simulator with a flexible pinna shall be used for all measurements. One recommended implementation is the
soft HATS pinna. It is an anatomically-shaped pinna which is structurally identical to the pinna described as Type
3.3 in ITU-T Recommendation P.57, except it has a hardness of 55, ±10 degrees Shore-OO (measured according to
ASTM 2240). (The Type 3.3 pinna has a hardness of 25, ±3 degrees Shore-A.) The other recommended
implementation is Type 3.4. The soft HATS pinna is recommended for all devices. Type 3.4 is recommended for
all devices except supra-concha and supra-aural headsets.
The soft HATS pinna and type 3.4 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
For information about other types of ear simulators, and the mouth simulator and test head with which they can be
used, see Annex B.
The soft HATS pinna and type 3.4 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. For most
measurements, the transformation from DRP to ERP shall be performed by using one of the tables in Annex C.
For measurements of distortion, the transformation from DRP to ERP shall properly take into account the frequency
of the measured distortion products relative to the stimulus frequency. This requirement may be fulfilled by using a
transformation filter as specified in Annex C. For measurement of distortion products occurring at specific
frequencies, such as harmonic distortion, this requirement may also be fulfilled by using a transformation table for
each individual distortion product.
For measurements of peak acoustic pressure, the transformation from DRP to ERP shall be fulfilled by using a
transformation filter as specified in Annex C.
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.
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5.2
IEEE P269/D9 Jan. 2002
Mouth simulators
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. This mouth is generally installed in a HATS.
ITU-T P.58 does not define a sound field behind the lip plane. However, practical experience has shown that this
mouth has a sound field distribution in the region between the HATS mouth and ear which 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.
For information about an alternative mouth simulator, and the ear simulators and test head with which they can be
used, see Annex B..
5.3
Test Fixtures
5.3.1
Test fixtures
The fundamental purpose of test fixtures is to test a handset or headset equipped device under conditions that most
closely approximate actual use by real persons. The test fixture shall be a HATS which complies with ITU-T
Recommendation P.58. When using the HATS soft pinna, the HATS shall also comply with ITU-T
Recommendation P.64, Annex E. When using Type 3.4, the HATS shall also comply with ITU-T Recommendation
P.64, Annex D.
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. (Similar effects
might be observed in sidetone and overall measurements.)
For information about an alternative test fixture, and the ear simulators and mouth simulator with which it can be
used, see Annex B.
5.3.2
Handset positioning
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.
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.
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.
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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.
A manufacturer may specify a recommended test position (RTP) on either the soft HATS pinna 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.
The final result shall be an “upper envelope” curve consisting of the maximum output of each measurement at each
frequency.
Insert new graph showing upper envelope with handset max pressure (13N) and default pos (6N or ERP)
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.
Handsets with carbon microphones require conditioning procedures before positioning for measurement. See Annex
D.
5.3.3
Headset positioning
Headsets should be tested in a position that most closely approximates real use. Natural headband pressure, or other
positioning techniques normally used be 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, may 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.
5.3.3.1
Headset microphone positioning
Three test positions are defined for the location of the headset microphone sound port, in order of preference.
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 ITU-T 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.
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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 on a fixed boom (for example, hanging on a wire), it may be necessary to average the
results using the procedure described for receivers in clause 5.3.3.2.
5.3.3.2
Headset receiver positioning
When a headset is placed on the soft HATS pinna or type 3.4 ear simulator, the 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 receive, sidetone and
overall measurements.
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 loudness rating is
1dB or less, and if the standard deviation of the frequency response is 2dB or less from 200 to 1000Hz, and 1dB or
less from 1000 through 4000Hz. If the results of the first 5 trials do not meet this criteria, report the results, but
label them as reduced accuracy.
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. A minimum of 5 measurements of all other
parameters should also be made. The mean results shall be reported.
For maximum acoustic output measurements made using the soft HATS pinna or 3.4 ear simulator, at least 5
additional measurements shall be made. The final result shall be an “upper envelope” curve consisting of the
maximum 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 1 and Figure 2.
130
L ERP (f)
120
110
100
90
100
1000
Frequency (Hz)
Figure 1 Maximum Acoustic Output, LERP(f), 5 measurements on one receiver
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10000
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130
L ERP(f)
120
110
100
90
100
1000
10000
Frequency (Hz)
Figure 2 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 shall apply.
5.4
Measurement microphones
Use free-field microphone (1/2 inch for test head and ¼ inch for HATS)for mouths, ½ inch pressure for room noise.
[Should encompass calibration microphones and measurement microphones, i.e. free-field, pressure, ½” diameter,
¼” diameter. Note, calibration section addresses using free-field and pressure microphones for mouth calibration.]
Use freq resp corrections when provided by mfgr. Deal w/ resolution issues.
Possible new words from Glenn Hess:
This clause addresses measurement microphones and calibration microphones. They include ½ inch diameter freefield, ½ inch diameter pressure, ¼ inch diameter free-field, and ¼ inch diameter pressure. Follow the
manufacturer’s recommendations on these microphones for frequency response and level corrections based on the
intended application.
Note, a ½ inch diameter microphone is used to calibrate a mouth simulator on a test head, and a ¼ inch diameter
microphone is used to calibrate a mouth simulator in HATS. The calibration clause addresses using free-field and
pressure microphones for mouth simulator calibrations.
5.5
Test environment
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
background noise level, and the test fixtures and device under test should be isolated from mechanical disturbances
that could cause significant error.
Be sure to recordthe test environmental conditions of temperature, humidity, and barometric pressure, in addition to
the background noise. Overall A-weighted noise level and 1/3 octave band sound levels are defined below.
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5.5.1
IEEE P269/D9 Jan. 2002
Background noise level
The background noise level in the test environment shall not exceed the limits shown in Table 1. The overall level
shall not exceed 29 dB A. However, these limits may be relaxed if it can be shown that the accuracy of the
measurement is not impaired.
Octave Band
Center Frequency (Hz)
63
125
250
500
1000
2000
4000
8000
Octave Band
Level
(dBSPL)
49
34
29
29
29
29
29
29
Table 1 Test room noise levels
5.5.2
Free field conditions
The test room shall have free field conditions that exist throughout the frequency range of interest. 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.
If an anechoic chamber is used as a test room to create a free field environment, it should be large enough to
comfortably accommodate the measurement transducers and the telephone.
5.5.3
Diffuse field conditions
1/3 octave only?
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.
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 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.
5.5.3.1
Test for diffuse field
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 10dB 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 2. The microphone is rotated about the DFTP through 360 degrees in each of three
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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.
Microphone rejection
at least
25dB or greater
20dB
15dB
10dB
Less than 10dB
Allowable variation
over 360 degrees
6dB
5dB
4dB
3dB
Microphone not suitable
Table 2 Diffuse field variation allowable vs microphone rejection
5.5.3.2
Test for spectrum uniformity
G(DFTP)(f) ( the spectrum at DFTP) shall be measured at DFTP and 6 additional points +/- 15cm from DFTP in each
of three mutually perpendicular planes. The spectrum at these 6 points shall not vary from G(DFTP)(f) by more than
+/- 3dB in each band.
Calibration of the diffuse field shall be according to 6.2.1 and 6.2.2, except performed at the DFTP.
5.5.4
Reference corner
The reference corner is one physical setup used for echo and stability tests. The reference corner consists of three
perpendicular plane, smooth, hard surfaces extending 0.5 m from the apex of a corner, as shown in Figure 3. The
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.
Should this be classified as an acoustic impairment and moved to the next clause? November meeting – Editor’s
choice. (Would make two sub-items under Impairments).
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250 mm
500 mm
Figure 3 – Reference corner for stability and echo tests
5.6
Acoustic impairments
In order to test an analog or digital telephone 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.
5.6.1
Hoth room noise
Hoth noise is random acoustic noise that has a power density spectrum corresponding to that published by Hoth.
This spectrum is designed to simulate typical ambient room noise. See Annex E for details. The noise level shall be
specified in dBA.
[Do we need other acoustic impairments, in principle one interfering voice, etc?? If not, no sub-clause JRB]
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Test Calibration
6.1
Measurement bandwidth and resolution
The calibration and measurement procedures shall be performed using the same bandwidth and measurement
resolution. The actual bandwidth or frequency interval and the measurement resolution shall be stated. See Annex F
for additional details. Also, review clauses F.9, F.11 and G.6 for a summary of the test signal parameters,
comparison of test methods and signals, and details about measurement bandwidth and resolution..
6.2
Send
6.2.1
Acoustic test spectrum
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. The electrical input to the mouth
simulator may be equalized to meet this requirement.
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. 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. Again, the electrical input to the mouth simulator may be equalized
to meet this requirement.
6.2.2
Acoustic test level
The standard test level for send, LMRP, is -4.7 dBPa at the MRP. Total harmonic distortion of the mouth simulator
shall be less than 2% for this test condition.
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 dBPa to this level.
6.2.3
Mouth simulator calibration procedure
[Re-write with HATS preferred, HATS figure. Settle ¼ inch min: pressure or free field.]
The preferred method is to calibrate at the MRP using a 12.5 mm (1/2 inch) free-field microphone oriented at 0
degrees to the mouth axis with the center of the protection grid at the MRP (Figure 1). 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.
A 6.25mm (¼ inch) free-field microphone may be used in place of a 12.5mm free-field microphone. In this case,
the 0.6dB correction is not applied.
An alternate method is to calibrate at the MRP using a 12.5mm pressure microphone oriented at 90 degrees to the
mouth axis with the center of the protection grid at the MRP. This method is valid only to 5 kHz.
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25 mm
Free Field
Microphone
MRP
Lip Ring of
Mouth Simulator
Figure 4 – On-Axis Calibration of mouth simulator
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.
The HATS mouth simulator shall be calibrated using one of the procedures in clause 6.2.3, unless a different
procedure is recommended by the manufacturer. (For example, a 6.25mm microphone at the MRP, oriented 90
degrees to the mouth axis.)
6.3
Receive and echo frequency response
Electrical test signals shall be applied from a 900 ohm resistive source impedance for 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.
A similar receive source calibration is required for testing handset and headset 4-wire devices. See clause 9.2 for the
source impedance requirements.
6.3.1
Electrical test spectrum
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. The electrical input may be equalized to meet
this requirement.
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. 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. Again, the electrical input may be equalized to meet this requirement.
6.3.2
Electrical test level
The standard electrical test level, nominal LRETP, is -16 dBV 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.
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The standard test level for digital telephones, nominal LRETP is –18.2dBV, which corresponds to –16dBm0. 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.
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 dBV to this level.
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Test Procedure for Analog Sets
Add parallel set coverage using circuit from TIA-470C draft.
7.1
General
Procedures are given in the following clauses for measurement of send, receive, sidetone, and overall performance
characteristics of handset and headset telephones. Parameters include frequency response, noise, input-output
linearity, distortion, mute, and maximum acoustic output. In addition, procedures are given for measuring telephone
set impedance and stability.
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 Volt per Pascal (dBV/Pa) for send,
decibels relative to one Pascal per Volt (dBPa/V) for receive, decibels relative to one Pascal per Pascal (dBPa/Pa)
for sidetone and overall, and decibels relative to one Volt per Volt (dBV/V) for echo. The stimulus level and signal
type for shall be reported 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.
7.1.1
Choice of test signals and levels
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 analysis methods and test signals.
The standard test signal for all telephones is artificial speech (F.6.1.1).
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 6.2.2 and 6.3.2.
7.1.2
Measurement bandwidth and resolution
The same bandwidth shall be used for calibration and measurements. The actual bandwidth used shall be stated.
Both calibration and measurement procedures shall be performed using the same measurement resolution. The
measurement resolution shall be stated.
In general, the test signals and analysis methods in this standard cover a frequency range of from approximately 100
to 8500Hz. The exact range depends on the analysis method, and perhaps also the test signal (see G.6)
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7.1.3
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Choice of ear and mouth simulators and test position
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.
7.1.4
Tone control setting
If the telephone is equipped with a tone control, the tone control shall be set to the manufacturer’s default setting.
This is the default tone control adjustment that shall be used for all measurements.
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.
7.1.5
Reference receive volume control setting
All measurements shall be done at the reference receive volume control setting (3.29). A range of volume control
settings may also be used where appropriate; such as, for example, minimum and maximum volume.
7.1.6
Reference send gain control setting
All measurements shall be done at the reference send volume control setting (3.30). A range of volume control
settings may be used where appropriate, such as minimum and maximum volume.
7.2
Analog DC Feed circuits
A general-purpose DC feed circuit is shown in Figure 5. 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:
C ≥ 50 microfarads
L ≥ 5 Henries (each)
R = 400 Ohms, including resistance of inductors
V = 50 Volts
A = measured current drawn by the telephone under test. Alternatively, the current can be fixed by a
current source, regardless of the R value.
[Should place these component values in the DC feed circuit figure.]
The feed circuit is terminated into a 900 Ohm resistive load. 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.
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C
receive electrical test point (RETP)
and
send electrical test point (SETP)
900 Ohms
C
analog
telephone
L
L
R
A
V
.
Figure 5 - General purpose DC feed circuit for 2-wire analog telephone
The loss of the feed circuit used should be measured. The loss should not be greater than 0.1dB over the range of
100 Hz to 8,000 Hz. The loss from 20 Hz to 100 Hz should not exceed 1 dB. The circuit of Figure 5, 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:
Feed circuit loss
= 20 log
Voltage across 900 Ohm resistor
Voltage across feed bridge
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.
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 6.
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C
C
L
L
L
L
Near-end
Analog
telephone
RN
Far-end
Analog
telephone
RF
AN
V
AF
V
Figure 6 - General purpose DC feed circuit for 2 analog telephones for overall measurements. (Subscripts
“N” and “F” refer to near-end and far-end, respectively.)
7.3
Analog telephone network impairments
Telephone performance can be influenced by various conditions in the network to which a telephone is connected.
The specific impairments described in clause 9.3 should be investigated. 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 Clause 9, but with the impairment introduced.
7.3.1
Loop current
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.
7.3.2
Network noise
Network noise can affect non-linear processes within an analog telephone. This shall approximate white noise with
levels measured in dBm, psophometrically weighted (dBmp), according to ITU-T Recommendation O.41 (1994).
7.3.3
Termination impedance
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.
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7.3.4
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Test loops
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. [Add the simulated loop circuits defined in TIA 470.]
7.3.5
Parallel sets
Extension telephones of varied DC and AC characteristics may be connected in parallel with the wireline analog
telephone under test. The telephone should be evaluated with a parallel set or equivalent to determine the effect on
acoustic performance. [Add the simulated parallel set circuit defined in TIA 470.]
7.3.6
Cordless range
A cordless telephone should be tested across the range of expected usage. This should include the minimum
specified distance the telephone is expected to operate between the base unit and mobile unit.
7.4
Receive
7.4.1
Receive frequency response
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 1 or Equation 2. HR(f) may be used to
calculate the receive loudness rating (RLR) according to ITU-T P.79-1999. Please see Annex H
H R ( f ) = 20 log
G ERP ( f )
G RETP ( f )
in dBPa / V
Equation 1
where:
GERP(f) is the RMS power spectrum at ERP
GRETP(f) is the RMS power spectrum at RETP
If the cross-spectrum method is used, the receive frequency response becomes:
H R ( f ) = 20 log
G( RETP )( ERP ) ( f )
G RETP ( f )
in dBPa / V
Equation 2
where:
G(RETP)(ERP) (f) is the cross spectrum.
7.4.2
Receive broad-band noise
Receive noise is internally generated audio frequency noise present at the telephone’s handset or headset receiver.
Measure the acoustic output signal at the ear simulator from 100 to 8,000Hz, averaging over a minimum period of 5
seconds. Receive noise should be measured with the telephone mute feature both “on” and “off.”
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The broad-band noise level is measured with “A” weighting in dBSPL (dBA). The measurement may be
implemented using single-channel FFT with Hann windowing or real-time spectrum analysis, followed by an “A”
weighted power summation, or an “A” weighting filter.
7.4.3
Receive narrow-band noise
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 32 Hz maximum from 100 to 8,000 Hz.
The receiver should be coupled to the ear simulator with the RETP terminated and with no signal input. Measure the
A-weighted receive noise level using a selective voltmeter or spectrum analyzer with an effective bandwidth of not
more than 32 Hz, over the frequency range of 100 to 8,000 Hz. If FFT analysis is used, then a “Flat Top”
windowing shall be employed.
7.4.4
Receive linearity
Receive linearity 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 sine wave signals are used, they shall be applied at the R10 frequencies from 200 through 5000 Hz, at 7 levels,
from –36 to –5dBV, in 5dB intervals. The reference stimulus level is –21dBV.
If artificial speech or another wideband stimulus is used, the test shall be performed at 7 levels, from –46 to –
16dBV, in 5dB intervals. The reference stimulus level is –16dBV. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
7.4.5
Receive distortion
Distortion tests for telephones are derived from standard measurement techniques defined in Annex J. Total
harmonic distortion has been traditionally used for analog telephones. However, continuous sine wave test signals
may not be suitable for telephones with digital signal processors (DSPs), automatic gain control, or compression
limiting. An alternative method is the difference-frequency distortion measurement also described in Annex J.
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 (+3dbV?). 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.
7.4.6
Receive mute Needs Re-working
Receive muting may be manually controlled or automatic, and can be activated via a mute feature execution, line
hold operation, touch-tone dialing, etc. Telephones equipped with a receive mute feature shall be measured for the
amount of acoustic muting provided. Receive mute may be measured in two ways. Mute attenuation is attenuation in
the speech path when the mute function is engaged. Mute interference is the amount of signal measured above the
muted noise floor when a stimulus is applied.
Mute may be measured with either of these methods using a single frequency measurement, multiple single
frequency measurements, or with speech or a speech-like signal. Care must be taken to assure that the frequencies
selected do not coincide with any unusual peak or dip in the normal receive frequency response.
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7.4.6.1
IEEE P269/D9 Jan. 2002
Mute Attenuation
Mute attenuation consists of measuring the receive frequency response as specified in clause 7.4.1 using the
standard –16 dBV input. The mute feature shall then be engaged and the test repeated at a stimulus level of 0dBV.
Mute attenuation can be calculated as either the difference in response levels from the unmuted condition to the
muted condition, or can be stated in absolute output levels for muted and unmuted states. Note, further calculations
will be needed to account for the two input levels used.
7.4.6.2
Mute Interference
Engage receive mute and measure the receive broad-band noise according to 7.4.2. Apply the test signal as listed in
clause 7.4.6.1 using 0dBV. Mute interference will be the relationship between telephone noise in the muted mode
and the muted stimulus signal.
7.5
Send
7.5.1
Send frequency response
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 3 or Equation 4. The send frequency response, HS(f) may be used to calculate the send loudness
rating (SLR) according to ITU-T P.79-1999. Please see Annex H
H S ( f ) = 20 log
G SETP ( f )
G MRP ( f )
in dBV / Pa
Equation 3
where:
GSETP(f) is the RMS power spectrum at SETP
GMRP(f) is the RMS power spectrum at MRP.
If the cross-spectrum method is used, the send frequency response becomes:
H S ( f ) = 20 log
G( MRP)( SETP) ( f )
G MRP ( f )
in dBV / Pa
Equation 4
where:
G(MRP)(SETP) (f) is the cross spectrum.
7.5.2
Send broad-band noise
Send noise is internally generated audio frequency noise present at the tip and ring terminals of the telephone.
Measure the electrical output signal at SETP, averaging over a minimum period of 5 seconds. Send noise should be
measured with the telephone mute feature both “on” and “off.” Send noise should be measured off-hook and onhook from 100 to 6500 Hz.
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Send broad-band noise is measured with psophometric weighting in units of dBm according to ITU-T
Recommendation O.41 (1994). This noise can be measured directly using a psophometrically weighted noise meter.
The measurement may be implemented using single-channel FFT with Hann windowing or real-time spectrum
analysis, followed by a psophometrically weighted power summation
7.5.3
Send narrow-band noise
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
narrow bands of not more than 32 Hz maximum from 100 – 6500 Hz.
With the receiver coupled to the ear simulator, measure the psophometrically-weighted noise level at the SETP with
a selective voltmeter or spectrum analyzer with an effective bandwidth of not more than 32 Hz, over the frequency
range of 100 to 6500 Hz. If FFT analysis is used, then a “Flat Top” windowing shall be employed.
7.5.4
Send linearity
Send linearity 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 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 +5.3dBPa, in 5dB intervals. The reference stimulus level is –9.7dBPa.
If artificial speech or another wideband test signal is used, the test shall be performed at 7 levels from –34.7dBPA to
–4.7dBPa, in 5dB intervals. The reference stimulus level is –4.7dBPa. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
7.5.5
Send distortion
Distortion tests for telephones are derived from standard measurement techniques defined in Annex J. Traditionally,
total harmonic distortion has been used for analog telephones. However, continuous sine wave test signals may not
be suitable for telephones with digital signal processors (DSPs), automatic gain control, or compression limiting. An
alternative method is the difference-frequency distortion measurement also described in Annex J.
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.
John Bareham to provide contribution to improve test procedure.
Steve Graham will investigate the mouth specifications to determine maximum frequency and level capability.
7.5.6
Send mute needs work
Send muting may be manually controlled or automatic, and can be activated via a mute feature execution for voice
privacy, line hold operation, touch-tone dialing, etc. Telephones equipped with a send mute feature shall be
measured for the amount of acoustic muting provided. Send mute may be measured in two ways. Mute attenuation is
attenuation in the speech path when the mute function is engaged. Mute interference is the amount of signal
measured above the muted noise floor when a stimulus is applied.
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Mute may be measured with either of these methods using a single frequency measurement, multiple single
frequency measurements, or speech or speech-like test signals. Care must be taken to assure that the frequencies
selected do not coincide with any individual frequency measured in the normal send response with a drastic
departure from a “smooth curve”.
7.5.6.1
Mute Attenuation
Mute attenuation consists of measuring the send frequency response as specified in clause 7.5.1 using the standard –
4.7 dBPa input. The mute feature shall then be engaged and test repeated at a stimulus level of 9.7dBPa.Mute
attenuation can be calculated as either the difference in response levels from the unmuted condition to the muted
condition, or can be stated in absolute output levels for muted and unmuted states. Note, further calculations will be
needed to account for the two input levels used.
[Rewrite as difference between 2 frequency responses]
7.5.6.2
Mute Interference
Engage send mute and measure the send broad-band noise according to 7.5.2. Apply the test signal as listed in
clause 7.5.6.1 using 9.7dBPa. Mute interference will be the difference in dB between the noise of the telephone in
muted mode and the muted stimulus signal.
[Rewrite ??]
7.5.7
Send frequency response in a diffuse field
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 5.
The diffuse field send frequency response may be sensitive to both the level and type of signal used.
H SD ( f ) = 20 log
G SETP ( f )
G DFTP ( f )
in dBV / Pa
Equation 5
where:
GSETP(f) is the RMS power spectrum at SETP
GDFTP(f) is the RMS power spectrum at MRP
The cross-spectrum method is not recommended.
7.5.8
Send signal-to-noise ratio
Send signal-to-noise ratio is a measure of the desired speech transmission relative to unwanted noise in the room
where the phone is used. See Annex K.
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7.6
7.6.1
IEEE P269/D9 Jan. 2002
Sidetone
Talker sidetone frequency response
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 6 or Equation 7. Talker
sidetone frequency response may be used to calculate the sidetone masking rating (STMR) according to ITU-T
P.79-1999. Please see Annex H
The STMR measured on an open-ear HATS is approximately 24dB. This represents the effective floor of STMR
measurements on actual telephones.
H TS ( f ) = 20 log
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 6
where:
GERP(f) is the RMS power spectrum at ERP
GMRP(f) is the RMS power spectrum at MRP
If the cross-spectrum method is used, the sidetone frequency response becomes:
H TS ( f ) = 20 log
G( MRP)( ERP ) ( f )
G MRP ( f )
in dBPa / Pa
Equation 7
where:
G(MRP)(ERP) (f) is the cross spectrum.
7.6.2
Listener sidetone frequency response
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 8.
H LS ( f ) = 20 log
G ERP ( f )
G DFTP ( f )
in dBPa / Pa
Equation 8
where:
GERP(f) is the RMS power spectrum at ERP
GDFTP(f) is the RMS power spectrum of the diffuse sound field in the room
The cross-spectrum method is not recommended.
This measurement is conducted using a uniform diffuse sound field as specified in clause 5.5.3.
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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..
7.6.3
Alternate method for listener sidetone
For the alternate method, listener sidetone response HLS(f) can be approximated by Equation 9. 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 9.
H LS ( f ) = H TS ( f ) − [ H S ( f ) − H SD ( f ) ] in dBPa / Pa
Equation 9
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.
7.6.4
Sidetone linearity
Sidetone linearity 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 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 +5.3dBPa, in 5dB intervals. The reference stimulus level is –9.7dBPa.
If artificial speech or another wideband test signal is used, the test shall be performed at 7 levels from –34.7dBPA to
–4.7dBPa, in 5dB intervals. The reference stimulus level is –4.7dBPa. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
7.6.5
Sidetone distortion
Distortion tests for telephones are derived from standard measurement techniques defined in Annex J. Traditionally,
total harmonic distortion has been used for analog telephones. However, continuous sine wave test signals may not
be suitable for telephones with digital signal processors (DSPs), automatic gain control, or compression limiting. An
alternative method is the difference-frequency distortion measurement also described in Annex J.
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.
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7.6.6
IEEE P269/D9 Jan. 2002
Sidetone delay
Sidetone delay is measured between the mouth simulator and the ear simulator, using one of the methods described
in Annex L
7.6.7
Sidetone echo response
If round trip sidetone delay is more than 5 ms, sidetone echo response should be measured. See Annex M.
7.7
Overall
[Does this refer to identical phones? Probably not. Do we need more words, or just leave as is?]
7.7.1
Overall frequency response
Overall frequency response is measured on two telephones connected back-to-back using the feed bridge shown in
7.7.1. This is a simulated end-to-end setup with no trunk loss 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 10 or Equation 11 below. It may be used to calculate the overall loudness rating (OLR) according to ITUT P.79-1999. Please see Annex H.
H O ( f ) = 20 log
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 10
where:
GERP(f) is the RMS power spectrum at ERP
GMRP(f) is the RMS power spectrum at MRP
If the cross-spectrum method is used, the overall frequency response becomes:
H O ( f ) = 20 log
G( MRP )( ERP ) ( f )
G MRP ( f )
in dBPa / Pa
Equation 11
where:
G(MRP)(ERP) (f) is the cross spectrum.
7.7.2
Overall linearity
Overall linearity 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.
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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 +5.3dBPa, in 5dB intervals. The reference stimulus level is –9.7dBPa.
If artificial speech or another wideband test signal is used, the test shall be performed at 7 levels from –34.7dBPA to
–4.7dBPa, in 5dB intervals. The reference stimulus level is –4.7dBPa. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
7.7.3
Overall distortion
Overall distortion is measured in a similar manner to sidetone distortion. However, this measurement is between two
telephone sets connected across a network connection.
7.8
Telephone set impedance
The ac voltage used in this measurement should have a magnitude representing typical voice signal values. A
suitable circuit is shown in Fig xx [Need Figure]To
7.8.1
AC impedance
The impedance, measured at the line terminals of the telephone set, should be determined over the frequency range
100 Hz to 5000 Hz.
Procedure??
7.8.2
Return loss
Need to include circuit, method, acoustic terminations (receive response, free-field, plane, corner), and ERL
calculation.
7.9
Stability test
Telephones can experience instability (howling or acoustic reverberation) 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 artificial 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
can be perceived as an audible howling or whistling from the telephone receiver, or repetitive fluctuation of the
telephone set line current.
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 3 of clause 5.5.4
.
7.10
Maximum acoustic output
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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.
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. 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.
7.10.1
Maximum acoustic pressure (long duration)
The maximum acoustic pressure is the maximum A-weighted, steady-state sound pressure emitted from a telephone
receiver. The stimulus for this test is a slow sweep applied at RETP. The measurement shall be made with an RMS
slow detector having 2-second effective averaging time, which is equivalent to a 1-second time constant.
A sine wave shall be swept from 100 Hz to 8,000 Hz with 10 Volts RMS at RETP. The sweep time shall be at least
80 seconds. A sweep time should be selected that provides consistent results, i.e. the result should be the same for a
test period ± 20 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.
7.10.2
Peak acoustic pressure (short duration)
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
unweighted “peak hold” level detector with a rise time equal to, or less than, 50 µs.
Connect the surge generator of Figure 7 to the battery feed (telephone side), so that terminal A is connected to the
positive terminal of the battery feed. Measure the peak pressure in the ear simulator while operating the surge
generator. An oscilloscope or a sound level meter, having an unweighted “peak hold” setting is used to make the
measurement. Reverse the telephone set connections and repeat.
Surge Generator
Figure 7 – Surge generator
[May want to move the surge generator text to the test equipment section.]
A surge generator meeting the following requirements shall be considered suitable for voltage stress testing to this
standard. However surge generators having different characteristics and component values may be required for
specific applications.
a) The short-circuit peak current shall not be less than 50 A.
b) The energy available at 1000 V is not less than 25 J.
c) The peak open-circuit voltage should not be less than specified.
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IEEE P269/D9 Jan. 2002
d) The open-circuit voltage waveforms produced by the surge generator shall meet the specified requirements.
e) These voltage waveforms include a 10 x 1000 ìs waveshape or 100 x 1000 ìs waveshape. [ *** Please
verify edit]
Circuit Parameters:
R1 = 1% resistor, 50 W
R2 = 1% resistor, 50 W
C = 1% capacitor, non-polarized, 1500 V
L = 1% inductor, resistance < 0.01 _
D = Reverse breakdown voltage > 1500 V (0.5 A)
S2 = Nonbounce switch, turn on delay < 1.0 µs, peak voltage > 1500 V, peak current > 50 A
NOTE: Component values are for ideal components. A real circuit would have to account for parasitic parameters of
the components used.
Operate the surge generator as follows:
a) Arrange the circuit elements to produce the desired waveshape.
b) Close S1 to charge the capacitor to the desired voltage.
c) Open S1.
d) Close S2 to fire the surge generator.
e) Open S2 and close S1 to recharge capacitor to the desired voltage.
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8
Test Procedures for Digital and 4-wire Systems
8.1
General
IEEE P269/D9 Jan. 2002
Use of a reference codec test methodology means that test procedures for digital telephone sets, in general, follow
those for analog telephone sets. The procedures in this clause assume a telephone set equipped with a handset or
headset. .
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 (dBPa/V) for
receive, decibels relative to one Volt per Pascal (dBV/Pa) for send, decibels relative to one Pascal per Pascal
(dBPa/Pa) for sidetone and overall, and decibels relative to one Volt per Volt (dBV/V) for echo. The stimulus level
and signal type shall be reported 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.
8.1.1
Choice of test signals and levels
We need to address frequency range for baseband vs wideband digital. Should either send or receive direction be
reduced in BW for baseband? Is standard BW (approx 100-8500) OK for all wideband tests?? JRB
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.
The standard test signal for all telephones is artificial speech (F.6.1.1)..
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 6.2.2 and 6.3.2.
8.1.2
Measurement bandwidth and resolution
The same bandwidth shall be used for calibration and measurement. The actual bandwidth used shall be stated. The
calibration and measurement shall be performed using the same measurement resolution. The measurement
resolution shall be stated.
In general, the test signals and analysis methods in this standard cover a frequency range of from approximately 100
to 8500Hz. The exact range depends on the codec, analysis method, and perhaps also the test signal (see G.6)
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8.1.3
IEEE P269/D9 Jan. 2002
Choice of ear and mouth simulators and test position
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.
For wideband applications, the type 1 ear simulator shall not be used, since it is intended for use only to 4,000Hz.
8.1.4
Tone control setting
If the telephone is equipped with a tone control, the tone control shall be set to the manufacturer’s default setting.
This is the default tone control adjustment that shall be used for all measurements.
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.
8.1.5
Reference receive volume control
All measurements shall be done at the reference receive volume control setting (3.29) A range of volume control
settings may also be used where appropriate, such as minimum and maximum volume.
8.1.6
Reference send gain control setting
All measurements shall be done at the reference send volume control setting (3.30). A range of volume control
settings may be used where appropriate, such as minimum and maximum volume.
8.2
Digital test circuits
8.2.1
Digital telephone interface
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 8. 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.
If digital test equipment is used, the digital telephone under test shall be connected using a direct digital interface as
shown in Figure 9. [Need to update figure.] 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.
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 10)
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IEEE P269/D9 Jan. 2002
Digital Translation
Interface
Linear PCM to Analog
Reference Codec
Analog
600 Ohms
RETP
SETP
Figure 8 – Analog interface to a digital set
[Editor – draw big box around two little box and label big box Reference Codec. Little Box on lieft is ADDA
andLittle Box on right is Digital Translation Coder]
Digital
Translation
Interface
Digital
Translation
Interface
Figure 9 – Digital interface to a digital set [Need to update figure.]
[Editor – Glenn Hess to provide sketch for updating this figure]
Digital
Translation
Interface
Digital
Translation
Interface
Figure 10 – digital interfacing for overall measurements
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8.2.2
Reference codec
8.2.2.1
General
IEEE P269/D9 Jan. 2002
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 CCITT Recommendation G.711. The codec
defined in this clause is based on that standard. For other coding schemes, an appropriate codec should be used.
8.2.2.2
Conversion Relationships
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.
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.
8.2.2.3
Other Parameters
In addition, reference codec characteristics, such as attenuation versus frequency distortion, idle channel noise, and
quantizing distortion should meet or exceed characteristics specified in CCITT Recommendation G.714 [6].
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 CCITT Recommendation O.133. Also see Supplement No. 21 to the
CCITT Blue Book, Vol. III.1. The intrinsic error of µ-law PCM encoding limits the signal-to-distortion ratio to
about 38 dB.
8.2.3
Wideband reference codec
8.2.3.1
General
There are a number of wideband codecs being used including CCITT 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.
8.2.3.2
Conversion Relationships
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.
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 ??? Coordinate with TIA 41.3.3. 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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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.
8.2.3.3
Other Parameters
In addition, wideband reference codec characteristics, such as frequency bandwidth and idle channel noise should
meet or exceed the characteristics specified below.
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 one of
the quiet codes or when the A/D digital output is connected to the D/A digital input.
8.3
Digital telephone network impairments
The most common digital impairments include delay, bit errors, frame or packet loss and transcoding. There are
many commercial units available to induce these impairments, and are usually specific to the type of digital
transmission system being tested.
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 for traditional ISDN
based systems are typically limited to speech compression transcoding (A-law to u-law conversions etc.), speech
path compression (G.726 ADPCM compression) and delay.
8.3.1
Network Delay
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. If the device delay is too long, then conversational dynamics could be
degraded once network delay is added.
8.3.2
Network Packet Loss
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, but it is
recommended that a pseudo subjective test be used such as PESQ (ITU-T P.832).
8.3.3
Echo Canceller
Network echo cancellers are typically deployed when network delay exceeds 25ms, one way. Echo cancellers can
affect non-linear speech path quality enhancing processes. It is recommended that the overall frequency response
and loudness rating be tested with network echo cancellers enabled and disabled, if applicable. There should be no
significant degradation of level or frequency response.
8.3.4
Discontinuous Speech Transmission
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. The 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.
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8.4
Receive
8.4.1
Receive frequency response
IEEE P269/D9 Jan. 2002
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 12 or Equation 13. The receive
response, HR(f), may be used to calculate the receive loudness rating (RLR), according to ITU-T P.79-1999. Please
see Annex H.
H R ( f ) = 20 log
G ERP ( f )
G RETP ( f )
in dBPa / V
Equation 12
where:
GERP(f) is the RMS power spectrum at ERP
GRETP(f) is the RMS power spectrum at RETP
If the cross-spectrum method is used, the receive frequency response becomes:
H R ( f ) = 20 log
G( RETP )( ERP ) ( f )
G RETP ( f )
in dBPa / V
Equation 13
where:
G(RETP)(ERP) (f) is the cross spectrum.
8.4.2
Receive broad-band noise
Connect the telephone set to the reference codec, and transmit idle code or silence to RETP. Couple the receiver to
the ear simulator and measure the output of the ear simulator. The telephone’s receiver should be isolated from
sound input and mechanical disturbances that would cause significant error. Receive noise should be measured with
the telephone mute feature both “on” and “off.”
The receive noise level is measured with “A” weighting in dBSPL (dBA). Measure the acoustic output signal at the
ear simulator, averaging over a minimum period of 5 seconds. The measurement may be implemented directly
using an “A” weighting filter, or by using single-channel FFT or real-time spectrum analysis, followed by an “A”
weighted power summation.
8.4.3
Receive narrow-band noise
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, which can then be compared to the overall weighted
background noise level.
With the receiver coupled to the ear simulator in a quiet environment (ambient noise less than 29 dBA), and with
idle code or silence at RETP, measure the A-weighted receive noise level at the ear simulator using a selective
voltmeter or spectrum analyzer with an effective bandwidth of not more than 31 Hz, over the frequency range of
100 to 8500 Hz. If FFT analysis is used, then “Flat Top” windowing shall be employed.
The same procedure applies for wide-band telephony applications.
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8.4.4
IEEE P269/D9 Jan. 2002
Receive linearity
Receive linearity 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 type used to measure frequency response.
If sine wave signals are used, they shall be applied at the R10 frequencies from 200 through 5000 Hz, at 7 levels,
from –38.2 to –8.2dBV, in 5dB intervals. The reference stimulus level is –23.2dBV.
If artificial speech or another wideband stimulus is used, the test shall be performed at 7 levels, from–48.2 to
−18.2dBV, in 5dB intervals. The reference stimulus level is –18.2dBV. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
8.4.5
Receive distortion
Total harmonic distortion using continuous sine wave test signals may not be suitable for digital telephones with
digital signal processors (DSPs), automatic gain control, or compression limiting. The preferred method is the
difference-frequency distortion measurement described in Annex J.
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.
8.4.6
Receive delay
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.
Receive delay is measured between RETP and the ear simulator.
One method to measure receive delay is to use 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 time capture type
measuring equipment (such as a digital storage oscilloscope) to acquire sufficient data for a clean measurement.
The input pulse is delivered to RETP, and triggers the time capture. Record the difference in time between the start
of the input pulse to RETP and the start of the measured pulse at the ear simulator.
Other methods for measuring delay are time delay spectrometry (TDS) and cross-correlation techniques. The delay
range of the measuring equipment must exceed the expected receive delay, or time aliasing may occur.
The input pulse is delivered to the RETP 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 receiver, as recorded by the ear simulator.
8.4.7
Receive out-of-band signals
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.
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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 8.5kHz (seeF.8).
The out-of-band signals shall be compared to the 1kHz. Signal level at the ear simulator.
(Steve & Glenn: Please check the above edits. JRB)
8.5
Send
8.5.1
Send frequency response.
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 14 or Equation 15. The send frequency response, HS(f) may be used to calculate the send
loudness rating (SLR) according to ITU-T P.79-1999. Please see Annex H
H S ( f ) = 20 log
G SETP ( f )
G MRP ( f )
in dBV / Pa
Equation 14
where:
GSETP(f) is the RMS power spectrum at SETP
GMRP(f) is the RMS power spectrum at MRP.
If the cross-spectrum method is used, the send frequency response becomes:
H S ( f ) = 20 log
G( MRP)( SETP) ( f )
G MRP ( f )
in dBV / Pa
Equation 15
where:
G(MRP)(SETP) (f) is the cross spectrum.
8.5.2
Send broad-band noise
Connect the telephone set to the reference codec and, in an active state with no acoustic input, measure the output at
SETP, averaging over a minimum period of 5 seconds. The telephone’s transmitter should be isolated from sound
input and mechanical disturbances that would cause significant error. Send noise should be measured with the
telephone mute feature both “on” and “off.” Send noise should be measured off-hook and on-hook.
The send noise level is measured with psophometric weighting in units of dBm according to ITU-T
Recommendation O.41 (1994). This noise can be measured directly using a psophometrically weighted noise meter,
or by using a single-channel FFT or real-time spectrum analysis, followed by a psophometrically weighted power
summation.
Send noise shall also be measured using the “A” weighting.
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8.5.3
IEEE P269/D9 Jan. 2002
Send narrow-band noise
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, which can then be compared to the overall weighted
background noise level.
With the receiver coupled to the ear simulator, mounted to the artificial head, measure the psophometricallyweighted noise level at the SETP, using a selective voltmeter or spectrum analyzer, with an effective bandwidth of
not more than 31 Hz, over the frequency range of 100 to 3500 Hz. If FFT analysis is used, then a “Flat Top”
windowing shall be employed.
The procedure shall then be repeated using “A” weighting instead of psophometric weighting, and the frequency
range shall be changed to 100 to 8500 Hz.
8.5.4
Send linearity
Send linearity 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 type used to measure frequency response.
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 +5.3dBPa, in 5dB intervals. The reference stimulus level is –9.7dBPa.
If artificial speech or another wideband test signal is used, the test shall be performed at 7 levels from –34.7dBPA to
–4.7dBPa, in 5dB intervals. The reference stimulus level is –4.7dBPa. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
8.5.5
Send distortion
Total harmonic distortion using continuous sine wave test signals may not be suitable for digital telephones with
digital signal processors (DSPs), automatic gain control, or compression limiting. The preferred method is the
difference-frequency distortion measurement described in Annex J.
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 wideband applications, the A weighting filter per ANSI S1.4 1983 shall be used instead of the psophometric
filter.
8.5.6
Send delay
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.
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.
One method to measure send delay is to use 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 time capture type
measuring equipment (such as a digital storage oscilloscope) to acquire sufficient data for a clean measurement.
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The input pulse is delivered to the mouth simulator and also triggers the time capture. Record the difference in time
between the start of the input pulse to the mouth simulator and the start of the measured pulse at SETP. The acoustic
delay between the mouth simulator and the microphone of the device should be considered negligible.
Other methods for measuring delay are time delay spectrometry (TDS) and cross-correlation techniques. The delay
range of the measuring equipment must exceed the expected send delay, or aliasing may occur.
8.5.7
Send out-of-band signals
Send 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.
Apply a sinewave signal at the MRP at a level of –4.7 dBPa, in the frequency range 300 to 3400 Hz. Measure the
signal level at SETP 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 SETP 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 8.5kHz (seeF.8).
The out-of-band signals shall be compared to the 1kHz. signal level at the SETP.
(Steve & Glenn: Please check the above edits. JRB)
8.5.8
Send frequency response in a diffuse field
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 16.
The diffuse field send frequency response may be sensitive to both the level and type of signal used.
H SD ( f ) = 20 log
G SETP ( f )
G DFTP ( f )
in dBV / Pa
Equation 16
where:
GSETP(f) is the RMS power spectrum at SETP
GDFTP(f) is the RMS power spectrum at MRP
The cross-spectrum method is not recommended.
8.5.9
Send signal-to-noise ratio
Send signal-to-noise ratio is a measure of the desired speech transmission relative to unwanted noise in the room
where the phone is used. See Annex K.
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8.6
IEEE P269/D9 Jan. 2002
Sidetone
Sidetone should be measured at minimum, nominal, and maximum volume settings.
8.6.1
Talker sidetone frequency response
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 17 or Equation 18. Talker
sidetone frequency response may be used to calculate the sidetone masking rating (STMR) according to ITU-T P.791999. Please see Annex H
The STMR measured on an open-ear HATS is approximately 24dB. This represents the effective floor of STMR
measurements on actual telephones.
:
H TS ( f ) = 20 log
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 17
where:
GERP(f) is the RMS power spectrum at ERP
GMRP(f) is the RMS power spectrum at MRP
If the cross-spectrum method is used, the receive frequency response becomes:
H TS ( f )
= 20 log
G( MRP)( ERP ) ( f )
G MRP ( f )
in dBPa / Pa
Equation 18
where:
G(MRP)(ERP) (f) is the cross spectrum.
8.6.2
Listener sidetone frequency response
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 19.
H LS ( f ) = 20 log
G ERP ( f )
G DFTP ( f )
in dBPa / Pa
Equation 19
where:
GERP(f) is the RMS power spectrum at ERP
GROOM(f) is the RMS power spectrum of the diffuse sound field in the room
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The cross-spectrum method is not recommended.
This measurement is conducted using a uniform diffuse sound field as specified in clause 5.5.3.
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.
8.6.3
Alternate method for listener sidetone
For the alternate method, listener sidetone response HLS(f) is given approximately by Equation 20. 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 8.6.1, measure the send frequency response per 8.5.1,
then measure the send frequency response in a diffuse field per 8.5.8, and apply Equation 20.
H LS ( f ) = H TS ( f ) − [ H S ( f ) − H SD ( f ) ] in dBPa / Pa
Equation 20
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.
8.6.4
Sidetone linearity
Sidetone linearity consists of measuring the talker sidetone frequency response as specified in Clause 10.6.1 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 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 +5.3dBPa, in 5dB intervals. The reference stimulus level is –9.7dBPa.
If artificial speech or another wideband test signal is used, the test shall be performed at 7 levels from –34.7dBPA to
–4.7dBPa, in 5dB intervals. The reference stimulus level is –4.7dBPa. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
8.6.5
Sidetone distortion
Total harmonic distortion measurements using continuous sine wave test signals may not be suitable for digital
telephones with digital signal processors (DSPs), automatic gain control, or compression limiting. The preferred
method is the difference-frequency distortion measurement described in Annex J.
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
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IEEE P269/D9 Jan. 2002
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.
8.6.6
Sidetone delay
Sidetone delay is measured between the mouth simulator and the ear simulator, using one of the methods described
in Annex L
8.6.7
Sidetone echo response
If round trip sidetone delay is more than 5 ms, sidetone echo response should be measured. See Annex M
8.7
Overall
8.7.1
Overall frequency response
[Note: 2 identical phones??]
The overall response should be measured using two telephone sets connected back-to-back, through the appropriate
digital telephone interface, with or without the ideal codec as necessary.
Overall frequency response is measured on two telephones connected back-to-back, using the interface shown in
Figure 10 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 21 or Equation 22. It may be used to calculate the overall loudness rating (OLR) according to ITU-T P.791999. Please see Annex H.
H O ( f ) = 20 log
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 21
where:
GERP(f) is the RMS power spectrum at ERP
GMRP(f) is the RMS power spectrum at MRP
If the cross-spectrum method is used, the overall frequency response becomes:
H O ( f ) = 20 log
G( MRP )( ERP ) ( f )
G MRP ( f )
in dBPa / Pa
Equation 22
where:
G(MRP)(ERP) (f) is the cross spectrum.
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8.7.2
IEEE P269/D9 Jan. 2002
Overall linearity
Overall linearity 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 type used to measure frequency response.
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 +5.3dBPa, in 5dB intervals. The reference stimulus level is –9.7dBPa.
If artificial speech or another wideband test signal is used, the test shall be performed at 7 levels from –34.7dBPA to
–4.7dBPa, in 5dB intervals. The reference stimulus level is –4.7dBPa. These levels take into account the high crest
factor of artificial speech, which approaches 20dB.
8.7.3
Overall distortion
Overall distortion can be measured in a manner similar to sidetone distortion, except that the measurement is made
from one set to another connected in a back to back configuration.
8.8
Echo frequency response
[JB: Add equivalent input noise procedure (as 6713)]
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 23 or
Equation 24. 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).
Should we reference terminally weighted coupling loss from 1329? As alternate? As preferred (as in 1329)?? JRB
H E ( f ) = 20 log
G SETP ( f )
G RETP ( f )
in dBV / V
Equation 23
where:
GSETP(f) is the RMS power spectrum at SETP
GRETP(f) is the RMS power spectrum at RETP
If the cross-spectrum method is used, the receive frequency response becomes:
H E ( f ) = 20 log
G( RETP)( SETP) ( f )
G RETP ( f )
in dBV / V
Equation 24
where:
G(RETP)(SETP) (f) is the cross spectrum.
Echo should be measured under the following four conditions:
a) Receiver placed on the same ear simulator used for receive measurements (10.3)
b) Handset or headset is suspended in an anechoic chamber, at least 500 mm from any reflecting objects
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c) Receiver and microphone facing a hard, smooth surface free of any object for 500 mm
d) In the reference corner of Figure 3 (5.5.4), 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 composite source signal (CSS, see F.7.1), without any filtering, and the
recommended test signal level is –12.2dBV (-10dBm0). This level results in a relatively good signal to noise ratio
for the measurement. The crest factor of CSS is less than 10 dB, allowing more headroom than artificial speech
according to P.50. For devices that incorporate non-linear processes, additional measurements using signal levels of
–28.2dBV (–26 dBm0) and –18.2dBV (–16 dBm0) may be performed. The measurement and calibration shall be
determined during the “On” portions of the signal. For wideband applications, the CSS test signal shall have a
bandwidth from 100 to 7000 Hz.
8.9
Stability loss
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 –12.2dBV (-10 dBm0) and less than or equal to –2.2dBV (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 23
or Equation 24). The measurement is performed under all four physical configurations specified in 8.8.
During the measurements, the operator should monitor the telephone for any sign of howling, whistling or other
signs of instability.
8.10
Convergence time
Some devices may have a nonlinear process to improve TCLw, such as an echo canceller. Such a system will have a
temporal characteristic for TCLw, such as convergence time. The convergence time is a measure of how fast the
full attenuation of the echo signal is achieved.
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 –12.2dBV is the preferred test signal for this measurement. (F.7.1)
8.11
Discontinuous speech transmission
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.
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.
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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.
8.11.1
General
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.
8.11.2
Measurement method
Telephone sets with adjustable receive levels shall be adjusted as close as possible to the reference nominal volume.
Use the same ear simulator and positioning which was used for receive measurements (8.4).
With both VAD disabled at the transmitting source and comfort noise generation on the receiving unit under test
turned off, a white noise test signal should be sent from the transmitting end such that the receive noise level
measured at the receiving telephone is 48 dBA. This test signal, at this level, will be assigned the level of ‘N dB’ as
a calibrated point for the purpose of the comfort noise test, since it may be generated either as an acoustic signal at a
‘golden’ transmitting telephone (and measured in dBA), or injected digitally (and measured in dBm0p).
If it is not possible to disable the VAD, then a band limited white noise signal at –60 dBm is input at RETP with a
10 dBm, 1kHz tone. The injected noise level is measured at the receiver, with the 1kHz tone filtered out. Remove
the 1kHz tone, and, once the device has discontinued the speech path, measure the generated comfort noise.
The following test sequence must be followed for all calibrated test noise levels of ‘M dB’ which will range from N10 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) 100-7000 Hz 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 2-4 are repeated for varying M in 1 dB gradations
8.12
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 with the telephone set in offhook conditions. See Annex N for a discussion of maximum pressure limits.
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. 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.
8.12.1
Maximum acoustic pressure (long duration)
The maximum acoustic pressure is the maximum steady state sound pressure emitted from a telephone receiver.
The stimulus for this test is a slow sweep applied at RETP. The measurement shall be made at the ear simulator
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with an RMS detector with a slow time constant (2-second effective averaging time, which is equivalent to a 1second time constant)..
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.
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 CCITT 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 80 seconds. A sweep time should be selected that provides consistent results, i.e.
the result should be the same for a test period ± 20 seconds.
8.12.2
Peak acoustic pressure ( short duration)
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
made at the ear simulator with an unweighted “peak hold” level detector 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 CCITT 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 11.
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250≤on≥500ms
IEEE P269/D9 Jan. 2002
≥ 500 ms
maximum positive
digital word
maximum negative
digital word
Figure 11 - 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.
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9
Test Procedures for Analog 4-wire Headsets and Handsets
9.1
General
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 maximum acoustic output and acoustic echo path loss.
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
referred to as “relative RLR” or “relative SLR”.
The headset shall be connected to the appropriate test circuit(s) described in clause 9.2. Other test circuits may be
used for specific applications. Records should be kept of the measurement conditions.
The measured frequency responses shall be presented as decibels relative to one Volt per Pascal (dBV/Pa) for send,
decibels relative to one Pascal per Volt (dBPa/V) for receive, and decibels relative to one Volt per Volt (dBV/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.
9.1.1
Choice of test signals and levels
In general, multiple test signals and stimulus levels should be used to ensure the headset or handset 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.
The standard test signal for all handsets and headsets is artificial speech (F.6.1.1).
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 6.2.2, and at the
receive stimulus level determined by the procedure in 9.3.2.
9.1.2
Measurement bandwidth and resolution
The same bandwidth shall be used for calibration and measurement. The actual bandwidth used shall be stated. The
calibration and measurement procedures shall be performed using the same measurement resolution. The
measurement resolution shall be stated.
In general, the test signals and analysis methods in this standard cover a frequency range of from approximately 100
to 8500Hz. The exact range depends on the analysis method, and perhaps also the test signal (see G.6)
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9.1.3
IEEE P269/D9 Jan. 2002
Choice of ear and mouth simulators and test position
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.
9.1.4
Tone control setting
If the handset or headset is equipped with a tone control, the tone control shall be set to the manufacturer’s default
setting. This is the default tone control adjustment that shall be used for all measurements.
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.
9.1.5
Default receive volume control and send gain adjustment
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 headsets and handsets 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.
9.2
Headset and handset test circuits
R1
Microphone
C
R2
+
V
Figure 12 Electret Microphone Test Circuit
Microphone
R2
Send
Electrical
Test Point
(SETP)
Figure 13 Dynamic Microphone Test Circuit
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Send
Electrical
Test Point
(SETP)
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IEEE P269/D9 Jan. 2002
C
R1
+
V
Microphone
R2
D
Send
Electrical
Test Point
(SETP)
L
Figure 14 Carbon Microphone Test Circuit
In the microphone test circuits, Figure 12, Figure 13 and Figure 14, the values for voltage V, capacitance C,
resistances R1 and R2, inductance L and diode D should equal the specified nominal 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 specified nominal impedance 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 13 may be used. The microphone should be connected to its intended host or suitable
simulation.
Receive
Electrical
Test Point
(RETP)
Z Ohms
Receiver or
Rceiver System
Figure 15 Receiver Test Circuit
The effective impedance ZS in the receiver test circuit of Figure 15 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 16 may be used for measurement of DC characteristics.
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A
DC
Ammeter
Microphone or
Receiver
R
V
DC
Voltmeter
+
V
Figure 16 DC Characteristics Test Circuit
9.3
Receive
9.3.1
General
Receive characteristics of handsets and headsets are measured with the receiver sound port terminated in the
appropriate ear simulator, as defined in Clause 5.
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.
9.3.2
Receive volume control adjustment
[Move to just below 9.1.4?? GH&JRB]
If a headset or handset is equipped with a receive volume control, it shall be set to the manufacturer’s default setting.
For frequency response measurements, LRETP shall be adjusted so that LERP = –14dBPa.
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 1kHz:
a)
Set the volume control to maximum. Adjust LRETP so that LERP = –14dBPa. Record this level and call
it LMAX.
b) Set the volume control to minimum. Adjust LRETP so that LERP = –14dBPa. If this is not possible,
move the control up slightly. Record this level and call it LMIN.
c)
9.3.3
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 = –14dBPa. This is the default receive volume control setting which shall be used for all
measurements unless otherwise specified..
Receive frequency response
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 25 or Equation 26.
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H R ( f ) = 20 log
G ERP ( f )
G RETP ( f )
IEEE P269/D9 Jan. 2002
in dBPa / V
Equation 25
where:
GERP(f) is the RMS power spectrum at ERP
GRETP(f) is the RMS power spectrum at RETP
If the cross-spectrum method is used, the receive frequency response becomes:
H R ( f ) = 20 log
G( RETP )( ERP ) ( f )
G RETP ( f )
in dBPa / V
Equation 26
where:
G(RETP)(ERP) (f) is the cross spectrum.
9.3.4
Receive broad-band noise
Receive noise is internally generated audio frequency noise present at the handset or headset receiver. Measure the
acoustic output signal at the ear simulator from 100 to 8,500Hz, averaging over a minimum period of 5 seconds.
Receive noise should be measured with the send mute feature both “on” and “off.”
The broad-band noise level is measured with “A” weighting in dBSPL (dBA). The measurement may be
implemented using single-channel FFT with Hann windowing or real-time spectrum analysis, followed by an “A”
weighted power summation or an “A” weighting filter.
9.3.5
Receive narrow-band noise
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 to 8,500 Hz.
The receiver should be coupled to the ear simulator with no input at Rx. Measure the A-weighted receive noise level
using a selective voltmeter, or a spectrum analyzer with an effective bandwidth of not more than 31 Hz, over the
frequency range of 100 to 8,500 Hz. If FFT analysis is used, then “Flat Top” windowing shall be employed.
9.3.6
Receive linearity
Receive linearity 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 sine wave signals are used, they shall be applied at
the R10 frequencies from 200 through 5000 Hz.
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 –15dB to LMID + 15dB, in 5dB intervals.
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9.3.7
IEEE P269/D9 Jan. 2002
Receive distortion
Distortion tests for handsets and headsets are derived from standard measurement techniques defined in Annex J.
Traditionally, total harmonic distortion has been used for handsets and headsets. However, continuous sine wave test
signals may not be suitable for handsets and headsets with automatic gain control or compression limiting. An
alternative method is the difference-frequency distortion measurement, also described in Annex J.
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 from –30 to 0 dBV (+3dbV?). 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.
9.3.8
AC impedance
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.
9.3.9
DC resistance
The resistance of the receive circuit should be obtained by the current-voltage method shown in Figure 16. This
measurement may be taken for various dc supply voltages, but use caution to avoid damaging the receive circuitry.
9.4
9.4.1
Send
Send gain control adjustment
[Move to just below 9.1.4?? GH&JRB]
If a headset or handset is equipped with a send gain adjustment, the gain control shall be set to the manufacturer’s
default setting. This is the default send gain control adjustment that shall be used for send frequency response
measurements.
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 1kHz:
a)
Set the gain adjustment to maximum. Set LMRP to –4.7dBPa, then measure LSETP. Record this level
and call it LMAX.
b) Set the gain adjustment to minimum. Set LMRP to –4.7dBPa, 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)
9.4.2
Calculate the halfway point in dB between LMAX and LMIN, and call it LMID. Set LMRP to –4.7dBPa,
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 spedcified.
Send frequency response
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 27 or Equation 28.
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H S ( f ) = 20 log
G SETP ( f )
G MRP ( f )
IEEE P269/D9 Jan. 2002
in dBV / Pa
Equation 27
where:
GSETP(f) is the RMS power spectrum at SETP
GMRP(f) is the RMS power spectrum at MRP.
If the cross-spectrum method is used, the send frequency response becomes:
H S ( f ) = 20 log
G( MRP)( SETP) ( f )
G MRP ( f )
in dBV / Pa
Equation 28
where:
G(MRP)(SETP) (f) is the cross spectrum.
9.4.3
Send overall noise
Send noise is internally generated audio frequency noise present at the microphone terminals or circuitry. Measure
the electrical output signal at SETP, averaging over a minimum period of 5 seconds. Send noise should be measured
with the mute feature both “on” and “off.” Send noise should be measured both off-hook and on-hook.
Send overall noise is measured with psophometric weighting according to ITU-T Recommendation O.41 (1994),
and also with A-weighting (dBA), in units of dBm. Psophometric measurements are made from 100-6500 Hz, while
A-weighted measurements are made from 100-8,500Hz.. These measurements can be made directly using a
psophometrically weighted or A-weighted noise meter. The measurement may be implemented using a singlechannel FFT with Hann windowing, or a real-time spectrum analysis, followed by a weighted power summation
JB: dBA units? DBm A-weighted? How to label: dBmA? Need to add purpose (predictive effect) for both
weightings and that you can use either one. In some cases, may want to use both weightings. Update here and in
clause 7 & 8.
9.4.4
Send narrow-band noise
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
narrow bands, of not more than 31 Hz maximum, from 100 – 6500 Hz.
With the receiver coupled to the ear simulator and mounted on the artificial head, measure the psophometricallyweighted 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 to 6500 Hz. If FFT analysis is used, then “Flat Top” windowing
shall be employed.
Should SFI also be done with A-weighting?? JRB
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9.4.5
IEEE P269/D9 Jan. 2002
Send linearity
Send linearity 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.
Sine wave signals shall be applied at the R10 frequencies from 200 through 5000 Hz for 7 levels, from –24.7 to
+5.3dBPa, in 5dB intervals. The reference stimulus level is –9.7dBPa.
The test shall be performed at 7 levels, from –34.7dBPA to –4.7dBPa, in 5dB intervals, for artificial speech. The
reference stimulus level is –4.7dBPa. These levels take into account the high crest factor of artificial speech, which
approaches 20dB.
9.4.6
Send distortion
Distortion tests for handsets and headsets are derived from standard measurement techniques defined in Annex J.
Total harmonic distortion has been traditionally used for handsets and headsets. However, continuous sine wave test
signals may not be suitable for handsets and headsets with automatic gain control or compression limiting. An
alternative method is the difference-frequency distortion measurement also described in Annex J.
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.
John Bareham to provide contribution to improve test procedure.
Steve Graham will investigate the mouth specifications to determine maximum frequency and level capability.
9.4.7
Send frequency response in a diffuse field
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 29.
The diffuse field send frequency response may be sensitive to both the level and type of signal used.
H SD ( f ) = 20 log
G SETP ( f )
G DFTP ( f )
in dBV / Pa
Equation 29
where:
GSETP(f) is the RMS power spectrum at SETP
GDFTP(f) is the RMS power spectrum at MRP
The cross-spectrum method is not recommended.
9.4.8
Send signal-to-noise ratio
Send signal-to-noise ratio is a measure of the desired speech transmission relative to unwanted noise in the room
where the phone is used. See Annex K.
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9.4.9
IEEE P269/D9 Jan. 2002
AC impedance
Connect the headset or handset receiver according to Figure 12, Figure 13 or Figure 14. 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.
9.4.10
DC resistance
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.
9.5
Echo frequency response
[JB: Add equivalent input noise procedure (as 6713)]
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 30 or
Equation 31. The inverse of this response is echo path loss.
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 headsets and handsets this calculation
shall be labeled as “relative TCLW,” since true TCLW is defined only for complete telephones.
TCLW may be normalized to nominal RLR and SLR target specifications. It shall then be labeled “normalized
TCLW,” and the method of normalization shall be stated.
H E ( f ) = 20 log
G SETP ( f )
G RETP ( f )
in dBV / V
Equation 30
where:
GSETP(f) is the RMS power spectrum at SETP
GRETP(f) is the RMS power spectrum at RETP
If the cross-spectrum method is used, the receive frequency response becomes:
H E ( f ) = 20 log
G( RETP)( SETP) ( f )
G RETP ( f )
in dBV / V
Equation 31
where:
G(RETP)(SETP) (f) is the cross spectrum.
Echo frequency response should be measured under the following four conditions:
a.
b.
Receiver placed on the same ear simulator used for receive measurements (10.3)
Handset or headset is suspended in the anechoic chamber at least 500 mm from any reflecting objects
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c.
d.
IEEE P269/D9 Jan. 2002
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 3 (5.5.4)Figure 1 (7.5.3), with the receiver placed 250 mm from the
corner
The recommended test signal for this test is composite source signal (CSS, see F.7.1), without any filtering, and the
recommended test signal level is LMID + 6dB. (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.)
9.6
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.
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. 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.
9.6.1
Maximum acoustic pressure (long duration)
The maximum acoustic pressure is the maximum steady state sound pressure emitted from a receiver. The stimulus
for this test is a slow sweep applied at RETP. The measurement shall be made with an RMS detector with a slow
time constant (2-second effective averaging time, which is equivalent to a 1-second time constant).
A sine wave shall be swept from 100 Hz to 10000 Hz for the two following conditions:
a) 10 dBV with a source impedance less than 10Ω
b) 15 dBV with a source impedance of 150Ω
The sweep time shall be at least 80 seconds. A sweep time should be selected that provides consistent results. That
is, the result should be the same for a test period ± 20 seconds.
9.6.2
Peak acoustic pressure (short duration)
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
the ear simulator with an unweighted “peak hold” level detector, with a rise time equal to or less than 50 µs.
Connect the surge generator of Figure 10, so that terminal A is connected to the positive terminal of the receive
circuitry. Measure the peak pressure in the ear simulator while operating the surge generator. An oscilloscope or a
sound level meter, having an unweighted “peak hold” setting is used to make the measurement. Reverse the
connection and repeat.
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Annex A
(normative)
Ear Simulators with Flexible Pinnas and Positioning Devices
A.1 General characteristics of the ear simulators
The soft HATS pinna 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.
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.
A.2 Differences between the two ear simulators
The soft HATS pinna is shaped like a real human ear, while Type 3.4 has a simplified shape.
The measured results obtained by the soft HATS pinna and type 3.4 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 ear simulators with a handset, the force applied to the type 3.4 pinna
will have to be between 2N and 10N more than that applied to the soft HATS pinna, with the soft
pinna currently available.
b) The acoustical input impedance of the two simulators is not identical. In general, the impedance of the
soft HATS pinna is slightly higher than that of the 3.4. For measurements with similar leakage, the
effect is that the receive loudness rating calculated from measurement on a soft HATS pinna could be
one to two dB lower (louder) than that obtained from the 3.4.
Regardless of these differences, both the soft HATS pinna and 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 soft HATS pinna and 3.4 is up to the user. However, measurements using the two simulators cannot be
expected to be exactly equal.
The recommendations in this standard for using the soft HATS pinna and 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.
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A.3 Handset Positioning devices
In principle, positioning of a handset on the soft HATS pinna or 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.
The positioning device currently available for the soft HATS pinna 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 currently available positioning device, placement by force on the soft HATS pinna 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.
The positioning device currently available for the type 3.4 ear 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 currently available positioning device 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 soft HATS pinna or
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.
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Annex B
(normative)
Alternative Ear Simulators, Mouth Simulator and Test Fixture
B.1 Alternative Ear Simulators
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, hard-cap, conically symmetrical receivers,
which naturally seal to the simulator rim, in the band of 100-4,000 Hz. These receivers should also be
tested in a realistic unsealed condition using the soft HATS pinna or type 3.4 as specified in this subclause.
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.
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Ear simulator recommendations are summarized in
Table B. 1:
Ear Simulator Types
Receiver Type
•
•
Supra-aural, hard cap
Supra-concha, hard cap
Type 2
•
Insert, sealed & unsealed
IEC 711
Type 3.1
•
Intra-concha
•
•
Supra-aural, hard cap
Supra-concha, hard cap
•
•
•
Intra-concha
Supra-aural
Supra-concha
•
•
•
Intra-concha
Supra-aural
Supra-concha
Type 1
IEC 318
Concha bottom simulator
Type 3.2
Simplified pinna simulator,
Low- or High- grade leak
Soft HATS pinna
Pinna
pinna
simulator
with
soft
(Recommended choice)
Type 3.4
Pinna simulator (simplified)
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
100-8,500Hz* bandwidth
6 N force (handset only)
100-8,500 Hz* bandwidth
•
•
6 N force (handset only)
100-8,500 Hz* bandwidth
(Recommended choice)
*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, the soft HATS pinna and type 3.4 ear simulators, DRP to ERP transformation shall be performed by
using one of the tables in Annex C. For type 3.2 ear simulators, DRP to ERP transformation shall be performed by
using the transfer function supplied by the manufacturer of the ear simulator.
For measurements of distortion, the transformation from DRP to ERP shall properly take into account the frequency
of the measured distortion products relative to the stimulus frequency. This requirement may be fulfilled by using a
transformation filter as specified in Annex C. For measurement of distortion products occurring at specific
frequencies, such as harmonic distortion, this requirement may be fulfilled by using a transformation table for each
individual distortion product.
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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.
B.2 Alternative Mouth Simulator
When an alternative ear simulator described in clause B.1 is used, an alternative mouth simulator may 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. The alternative mouth simulator is 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 P.58. There are minor differences between these
specifications, so there may be small differences between the simulators.
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.
B.3 Alternative Test Fixture
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. (This has sometimes been referred to as LRGP-H.)
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.
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Annex C
(normative)
DRP TO ERP Corrections
The ear simulators in HATS are made in such a way that measurements are made at the drum reference point (DRP).
For telephony measurements, the ear reference point (ERP) is used to maintain comparability to measurements on
handsets and headsets, which are measured at the ERP.
The DRP to ERP correction in this annex is from ITU-T Recommendation P.57 (1996).
The DRP to ERP correction SDE must be added to the data measured at the DRP in order to correct to the ERP. The
effect is to remove a broad frequency response peak of about 10dB in the region of 3000Hz.
DRP to ERP Correction (SDE)
5
Amplitude (dB)
0
-5
-10
-15
-20
-25
100
1000
Frequency (Hz)
Figure C. 1 1/12 Octave Filter Center Frequencies
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10000
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Frequency
(Hz)
92
97
103
109
115
122
130
137
145
154
163
173
183
193
205
218
230
244
259
274
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
IEEE P269/D9 Jan. 2002
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
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77
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
For IEEE/STIT work only. Do not submit to any other standards organization.
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
IEEE P269/D9 Jan. 2002
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 ISO R40 Preferred Frequencies
For maximum acoustic pressure (long duration) measurements, transformation from DRP to ERP shall be done
using Table C. 1 or Table C. 2. Transformation to free field or diffuse field shall be made using the transfer function
supplied by the manufacturer of the ear simulator, if available. Alternatively, the transfer functions specified in ITUT 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
For peak acoustic pressure (short duration) measurements, the transformation shall be implemented using a
minimum phase parametric filter or equivalent, since peak measurements must 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.
The filter parameters for transformation from DRP to ERP shall be based on the transfer functions in this Annex.
The filter parameters for transformation to free field or diffuse field shall be made using the transfer function
supplied by the manufacturer of the ear simulator, if available. Alternatively, the transfer function specified in ITUT Recommendation P.58 can 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
The magnitude of the filter response shall follow the transfer function within a tolerance of +/-2dB.
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Annex D
(normative)
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.
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.
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Annex E
(normative)
Hoth Room Noise
Hoth noise can be described as acoustic random noise that has a power density spectrum corresponding to that
published by Hoth [need new bibliographic reference]. The spectrum of Hoth noise is designed to simulate typical
ambient room noise over time.
Test Table
1m
Loudspeaker
Plan View
Loudspeaker
50cm
Test Table
Side View
Figure E. 1 Hoth noise test setup
Hoth noise can be reproduced using two non-correlated white noise generators and two equalizers in order to
produce the required spectrum through four loudspeakers positioned radially 50 cm above the table, 1 meter away
from HFRP and 45 degrees apart (see Figure E. 1). Each one of the two uncorrelated noise signals are delivered to
two loudspeakers in alternated fashion.
Using a free field microphone placed at the HFRP in absence of the test table, the 1/3 octave spectrum can be
calibrated. Once the spectrum is within ± 2 dB in each band of the Hoth specification, replace the table. The overall
A weighted level can now be set with a probe microphone located close to the microphone on the telephone, with
the probe microphone configured to measure dBSPL with A weighting (dBA).
Table E. 1 below gives the spectrum density adjusted in level to produce a reading of 50 dBA. Figure E. 2 shows a
plot of this spectrum. The spectrum below is independent of level and shifts equally for each 1/3 octave band.
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Frequency (Hz)
Spectrum Density Bandwidth
(dB SPL/Hz)
_ƒ (dB)
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
IEEE P269/D9 Jan. 2002
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
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Hoth Noise Spectrum Density Vs Frequency
Spectrum Density (dBSPL/Hz)
35
30
25
20
15
10
5
0
-5
-10
100
1000
Frequency (Hz)
10000
Figure E. 2 Hoth noise spectrum
Typical Hoth noise levels range from 35 dBA to 65 dBA. Switching parameter and speech detection tests should be
performed in this range in 10 dB increments.
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.
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Annex F
(normative)
Test Signals
F.1 General
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.
F.2 Classifications
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.
In addition to the signals described in Annex F, signals described in ITU-T P.501 are also recommended when they
are appropriate.
F.3 Modulation types
Several types of modulation may be applied to deterministic or random signals.
approximate the syllabic rhythm of real speech.
This is done in order to
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).
F.3.1
Square wave modulation
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 international telephone testing methods. It is close to the modulation rate of real
speech. Other timing patterns may be used after confirming that the maximum measured response has been reached.
In some cases, a periodic pulse pattern of this type will not correctly activate the telephone circuit. In such cases, a
randomly varied pulse pattern may be used. The average “on” and “off” times should approximate 250 ms and 150
ms respectively.
With this type of modulation, all measurements are to be performed during the “on” part of the pattern. For other
types of modulation, the signal is to be measured during the entire presentation time.
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F.3.2
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Sine wave modulation
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.
F.3.3
Pseudo-random modulation
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.
F.4 Deterministic signals
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.
F.4.1
Sine wave
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.
F.4.2
Pseudo-random
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 to be 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-random signals. If square wave modulation is used, the “on” time must
correspond to one or more complete period(s) of the pseudo-random signal.
F.5 Random signals
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.
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.
F.5.1
White noise
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.
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F.5.2
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Pink noise
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.
F.6 Speech-like signals
Speech-like signals include ITU-T Recommendation P.50 (1993) artificial voice, 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.
F.6.1
Simulated speech
Typical parameters of simulated speech include long-term average spectrum, short-term spectrum, instantaneous
amplitude distribution, speech waveform structure, and the syllabic envelope.
F.6.1.1
P.50 Artificial Voice.
ITU-T Recommendation P.50 (1993) defines the temporal and spectral parameters for a test signal which emulates
the characteristics of speech. This artificial voice is a continuous speech signal with a frequency range of 89.1 Hz to
8919 Hz. Pauses may need to be inserted to emulate the on-off characteristics of conversational speech. See 6.6.1.2
for information on inserting pauses.
F.6.1.2
P.59 Artificial Conversational Speech.
Artificial conversational speech is a test signal generated by inserting pauses in the continuous artificial voice signal
described by ITU-T Recommendation P.50 (1993). 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.
F.6.1.3
Simulated Speech Generator (SSG).
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 O for details of this signal.
F.6.2
Synthesized speech
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.
F.6.3
Real speech
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.
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F.7 Compound signals
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
measurement. By applying two signal sources, one can be used specifically for “biasing” the unit into a stable,
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.
F.7.1
Sequential presentation
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.
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.
If the signal includes pauses, calibration and measurements are to be performed during the “on” part of the pattern.
F.7.2
Simultaneous presentation
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.
F.7.2.1
TDS Sweep with P.50 Noise Bursts.
This compound signal has two components, which are presented at the same time, but not synchronized with each
other. The bias signal (P.50 noise burst) is intended to insure that the telephone is in a stable, well-defined operating
state. The measurement signal (TDS sweep) 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. See Annex P for a detailed description of this signal.
F.7.2.2
TDS Sweep with P.50 Artificial Voice.
Similar to F.7.2.1, except the bias is a continuous artificial voice signal defined in ITU-T Recommendation P.50
(1993). (See F.6.1.1).
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TDS Sweep with Real Speech
Similar to F.7.2.1, except the bias is real speech (F.6.3).
F.7.2.4
TDS Sweep with Random or Pseudorandom Noise
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.
F.7.2.5
Pseudorandom Noise with P.50 Noise Bursts.
This compound signal has two components, which are presented at the same time, but not synchronized with each
other. The bias signal (P.50 noise burst) 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.
F.7.2.6
Pseudorandom Noise with P.50 Artificial Voice.
Similar to F.7.2.5, except the bias is a continuous artificial voice signal defined in ITU-T Recommendation P.50
(1993). (See F.6.1.1).
F.7.2.7
Pseudorandom Noise with Real Speech
Similar to F.7.2.5, except the bias is real speech (F.6.3).
F.7.2.8
Pseudorandom Noise with Random or Pseudorandom Noise
Similar to F.7.2.5, 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.
F.7.2.9
Sine Wave with Notched Real Speech.
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.
F.8 Test signal bandwidth
In general, the test signals and analysis methods in this standard cover a frequency range of from approximately 100
to 8500Hz. 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-toERP correction curve (Annex C). For digital phones, the exact range may also be determined by the codec.
Some signals, such as SSG (F.6.1.3), are defined only for a smaller bandwidth, and cannot be used outside their
defined range.
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F.9 Signal parameter summary
Table 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.
Usable
Bandwidth
(Hz)**
Document Test Signal
Ref.
Maximum
Analysis
Interval
Alternative
Analysis Format
F.4.1
Sine Wave*
100-8,500
ISO R40 steps
1/12 Oct. steps
F.4.2
Pseudo-Random*
100-8,500
25 Hz bands
1/12 Oct. bands
F.5.1
White Noise*
100-8,500
25 Hz bands
1/12 Oct. bands
F.5.2
Pink Noise*
100-8,500
1/12 Oct. bands
25 Hz bands
F.6.1.1
P.50 Artificial Voice
100-8,500
1/12 Oct. bands
25 Hz bands
1/12 Oct. bands
25 Hz bands
100-5,000
1/12 Oct. bands
25 Hz bands
F.6.1.2
F.6.1.3
P.59
Artificial
Conversational Speech
Simulated
Speech
Generator
100-8,500
F.6.2
Synthesized Speech
100-8,500
1/12 Oct. bands
25 Hz bands
F.6.3
Real Speech
100-8,500
1/12 Oct. bands
25 Hz bands
F.7.1
Composite Source Signal
100-8,500
25 Hz bands
1/12 Oct. bands
TDS Sweep with Bias
100-8,500
50 Hz Bands
1/12 Oct. bands
Pseudorandom noise with
Bias
100-8,500
50 Hz Bands
1/12 Oct. bands
F.7.2.1F.7.2.4
F.7.2.5F.7.2.8
* Modulation may be required depending on the application. See F.3.
** Nominal bandwidth. See G.6.
Table F. 1Test signal parameters
F.10 Test signals published on CD-ROM
The artificial voice according to ITU-T P.50 , as well as a large speech database, is included on a CD-ROM
published as ITU-T 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 P.501
.
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F.11 Signal and test method comparative summary
Table 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.
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*
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*
5.2.1
5.2.2
5.2.3
5.3.1
5.3.2
5.4.1
FFT/Cross Spectrum
Dual-Channel FFT
Single-Channel FFT
Max. Length Seq.
Real-Time Filter
Dual-Channel RTA
Single-Channel RTA
Sine-Based
Discrete Tone
Table F. 2 Signal compatibility with test method
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Annex G
(normative)
Analysis Methods
G.1 General
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.
G.2 Fast Fourier transform (FFT) and cross spectrum analysis
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.
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 determines 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, the
averaging time shall be at least one full period of the signal. Averaging time shall be stated for all measurements.
G.2.1
Dual-channel FFT
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.
G.2.2
Single-channel FFT
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.
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G.2.3
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Maximum length sequence (MLS) analysis
The MLS technique (Rife and Vanderkooy [B1]) 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 noncoherent power. Some non-linear analysis capabilities and post-processing are available. This method can improve
measurement S/N.
G.3 Real-time filter analysis (RTA)
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 S1.11-1986 (R 1998) 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).
When analyzing a periodic signal such as pseudo-random noise or a segment of real or artificial speech, the
averaging time shall be at least one full period of the signal. Averaging time shall be stated for all measurements.
G.3.1
Dual-channel real-time filter analysis
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.
G.3.2
Single-channel real-time filter analysis
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
telephone under test to be time invariant, and is limited in measuring harmonic distortion. Some direct postprocessing of the data may be possible.
G.4 Sine-based analysis
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.
G.4.1
Discrete tone (stepped sine)
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 S1.6-1984 (R1997))
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
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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.
G.4.2
Swept sine
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.
G.4.3
Time delay spectrometry (TDS)
TDS (Heyser [B2]) 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 filter and a detector. 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.
G.5 Simulated free field techniques
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.
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.
The frequency resolution available in an anechoic chamber is largely determined by the measurement technique
employed. Anechoic chambers are limited at low frequencies by the size of the open space available and the depth
of the absorptive material on the walls, floor and ceiling.
G.6 Measurement bandwidth and measurement resolution
In general, the test signals and analysis methods in this standard cover a frequency range from approximately 100 to
8500Hz. The lower limit is determined by the mouth simulator, whose practical lower limit is approximately
100Hz. for general use. The upper limit is determined by the range of the DRP-to-ERP correction curve (Annex C).
The exact range depends on the analysis method, and, in some cases, the test signal. (Some signals, such as SSG
(F.6.1.3), are defined only for a smaller bandwidth, and cannot be used outside their defined range.)
For example, if continuous-spectrum signals such as artificial speech are analyzed in 1/12th octave bands, the range
includes the bands centered from 103 through 8660 Hz. NOTE: We might have to go down to the 92Hz Band,
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since P.50 is ndefined in 1/3 octaves to 100 Hz. If continuous-spectrum signals such as the noise part of CSS are
analyzed in linear format, the range includes the lowest band whose lowest edge is approximately 100 Hz. NOTE:
Might have to go to 90Hz. , through the band whose center frequency is approximately 8500Hz. For sinewaves, the
range is from 100 through 8500 Hz.
The standard frequency pattern for sinusoidal signals is the R40 sequence, from 100 through 8500 Hz. (See Table G.
1 and Table G. 2). However, when testing digital devices, or devices which have any internal digital processing,
some of these frequencies should be adjusted up to +/- 1% (Glenn to check this tolerance) 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.
FOR DISCUSSION & ACTION: Should there be some statement about phones, e.g. narrowband digital, which
simply cannot operate above 4kHz? Should electrical stimulus be limited to 4kHz? What about the receiver
measurement? What about the mouth signal? Do people talk in a different bandwidth or level when using a
4kHz phone?? JRB This is a wideband standard except for type 1 ears and 8 kHz sampling rate digital phones –
put in scope? Work on centralizing references to bandwidth and resolution.
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. 1 and Table 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. 1. The frequencies are
actually based on 10 bands per decade.
f = 10 (n / 10 )
Equation G. 1
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. 2.
f = 10 (n / 10) ± 0.05
Equation G. 2
Example: For the 100 Hz band, the 1/3 octave band number = 20. The exact center frequency is 100 Hz, the lower
limit is 89.13Hz, and the upper limit is 112.20Hz.
For the 125 Hz band, the band number = 21. The exact center frequency is 125.89Hz, the lower limit is 112.20Hz,
and the upper limit is 141.25Hz.
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. 3.
f = 10 (n + 0.5 ) 40
Equation G. 3
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Example: 1/12 octave band number 80 has a center frequency of 102.92Hz.
The 1/12 octave passband upper & lower limit frequencies can be calculated according to Equation G. 4.
f = 10 [(n + 0.5 ) 40] ± 0.0125
Equation G. 4
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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
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.
IEEE P269/D9 Jan. 2002
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
IEEE P269/D9 Jan. 2002
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
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Annex H
(normative)
Loudness Rating Calculations
ISO R10 format data is required for calculating loudness ratings according to ITU-T P.79. Measured frequency
rersponses (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
using “band-averaging”. The measured points within a particular 1/3 octave band are “power averaged” according to
Equation H. 1, and assigned to theR10 frequency at the band center.
At each ISO R10 preferred frequency
1
H ′ ( f ) = 10 log 10 
N
N
∑ 10
i =1
Hi
10



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. 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.6.
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.
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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.
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Annex I
(normative)
Linearity
Linearity 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 1dB, 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 R40, might be useful.
Linearity shall be measured using the same stimulus type used to measure frequency response (send, receive,
sidetone, or overall). The test shall be at 7 levels, in 5dB intervals. The reference stimulus level shall be specified.
Each individual measurement is processed according to equation Equation 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, the result is 7 parallel lines at levels from –15 to +15dB (see Figure I. 1). Nonlinearities are
displayed as variations from the parallel lines (see Figure I. 3 through Figure I. 6).
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20.00
+15
dB
+10
10.00
+5
Reference
Stimulus
-4.7dBPa
0.00
-5
-10.00
-10
-15
-20.00
100
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 1 Essentially linear phone
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.
+15
Output
dB
0.00
-15
-15
-10
-5
0
+5
+10
+15
Input, dB
Figure I. 2 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 Figure I. 1 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 Figure I. 2. In figure Figure
I. 1, the same information is displayed at all frequencies in one graph.
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20.00
+15
dB
+10
10.00
+5
Reference
Stimulus
-4.7dBPa
0.00
-5
-10.00
-10
-15
-20.00
100
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 3 Receiver with compression at low-frequency resonance
20.00
dB
+15
10.00
+10
+5
Reference
Stimulus
-4.7dBPa
0.00
-5
-10
-10.00
-15
-20.00
100
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 4 Wideband compressor with 1.5 to 1 ratio
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+15
30.00
dB
+10
20.00
+5
10.00
Reference
Stimulus
-4.7dBPa
0.00
-5
-10.00
-10
-20.00
-15
-30.00
100
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 5 Midband expander with approximately 1 to 2 ratio
20.00
+15
dB
+10
10.00
+5
Reference
Stimulus
-4.7dBPa
0.00
-5
-10.00
-10
-15
-20.00
100
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 6 High frequency limiting due to overload in pre-emphasis circuit
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Annex J
(normative)
Distortion Work: sines vs narrowband random; default method?
J.1
Total harmonic distortion
Distortion is a measure of nonlinearities. Signals appear at the output of a device at frequencies not present in the
input. Distortion is also nonlinear in the sense that it 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.
Several standard distortion measurement methods use sinusoidal stimulus: harmonic, harmonic + noise, and
difference-frequency distortion. These are defined in Clause J.2.
(Another standard method, intermodulation distortion, 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.)
Distortion test methods using sines may not be suitable for use on many telephones. Extensions of sine methods are
described.
The recommended method is signal-to-distortion-and noise ratio (SDN), defined in Clause J.4. It uses a narrowband pseudo-random noise as the stimulus, and analysis of THD + noise with a weighted notch filter.
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.
J.1.1
Test signal
Three types of distortion test signals are recommended. These include continuous sine waves, modulated sine
waves, and narrow-band pseudo-random noise. A square wave, sine wave, or a pseudo-random modulation, can be
used to modulate the sine wave signals. Refer to Clauses F.3 and F.4.
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.1.2
Suitability test
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 ±2dB of the comparable values previously obtained in the complete frequency response
measurement, then the proposed distortion test signal is suitable. Distortion does not have to be measured using the
same test signal as is used for measuring frequency response.
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J.2
J.2.1
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Sinusoidal Methods
Total harmonic distortion (THD)
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. 1 - Equation J. 4.
Harmonic analysis may be done using bandpass filters, or an equivalent algorithm.
% THD
= 100
power sum of included harmonics
fundamental
Equation J. 1
= 100
% THD
( A2 ) 2 + ( A3 ) 2 + . . . + ( An ) 2
A1
Equation J. 2
Alternatively,
% THD
= 100
power sum of included harmonics
unfiltered total output
Equation J. 3
% THD = 100
( A2 ) 2 + ( A3 ) 2 + . . . + ( An ) 2
( A1 ) 2 + ( A2 ) 2 + ( A3 ) 2 + . . . + ( An ) 2
Equation J. 4
Where
An = amplitude of nth product
J.2.2
Total Harmonic Distortion (THD) and noise
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. 5 and Equation J. 6.) It is usually expressed as a percent.
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.
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% THD + Noise = 100
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output from notch filter
unfiltered total output
Equation J. 5
% THD + Noise = 100
( A2 ) 2 + ( A3 ) 2 + . . . + ( An ) 2 + ( Anoise ) 2
( A1 ) 2 + ( A2 ) 2 + ( A3 ) 2 + . . . + ( An ) 2 + ( Anoise ) 2
Equation J. 6
Where
An = amplitude of nth product
Anoise = amplitude of wideband noise and nonharmonic products
J.2.3
Difference-frequency distortion (DF Distortion)
Difference-frequency distortion is measured by using two stimulus signals, typically spaced from 20 to 200Hz apart.
A complex group of distortion products results, consisting of odd and even order products. (Equation J. 7 and
Equation J. 8) 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) lie above the set’s cutoff frequency.
% Total DF Distortion
= 100
power sum of included products
power sum of both stimulus signals
Equation J. 7
% Total DF Distortion = 100
( A−2 ) 2 + ( A3 ) 2 + ( A−3 ) 2 + ( A−4 ) 2 + ( A5 ) 2 + ( A−5 ) 2 + . . .
( A f 1)2 + ( Af 2 )2
Equation J. 8
Where
An = amplitude of nth product
Afn = amplitude of nth stimulus signal products
J.3
Alternate stimulus signals
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.
The most simple alternative is to use modulated sine waves as the stimulus. A square wave, sine wave, or a pseudorandom modulation can be used to modulate the sine wave signals. Refer to Clause F.3 for details. The analysis is
the same as for continuous sinewave methods.
Another alternative may be suitable when the device to be tested has extensive signal processing based on speech.
For harmonic distortion measurements, a narrow-band pseudo-random noise signal is the stimulus. For differencefrequency distortion, two narrow-band pseudo-random signals are the stimulus. The stimulus level is calculated on a
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power basis. Analysis is with a notch filter for THD + noise, and with bandpass filters, or an equivalent algorithm,
for THD or DF distortion.
J.4
Recommended method: signal-to-distortion-and noise ratio (SDN)
The default distortion test method for this standard is to use a narrow-band pseudo-random noise as the stimulus,
and analyze THD + noise with a weighted notch filter. See Equation J. 9 and Equation J. 10.
Narrow-band pseudo-random noise (F.4.2) should be used as the default test signal for all distortion measurements.
It should have an effective bandwidth of 25 to 50 Hz, and 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. The crest factor
should be 9±3 dB. When a continuous narrow-band pseudo-random test signal is not suitable, modulation may be
applied in a similar manner to modulating a sine wave test signal.
The output fundamental is measured with a bandpass filter or equivalent algorithm. Send distortion is measured
using the psophometric filter in ITU-T Recommendation O.41 (1994), but with a notch added to eliminate the test
signal. Receive distortion is measured using an A-weighting filter according to ANSI S1.4-1983 (R1997) with the
notch as well.
Output from the notched filter includes harmonics 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. 9 and Equation J. 10. The
result is signal-to-total distortion and noise ratio, psophometrically or A-weighted depending on the measurement.
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.
The filter shall compensate for the notch on a power basis. A constant shall be added to each point of the notched
filter frequency response so that the power sum of all points, on a logarithmic frequency scale, is equal to the power
sum of all frequency response points of the original psophometric or A-weighted filter.
% SDN
= 100
output from weighted notch filter
narrow − band noise stimulus
Equation J. 9
% SDN
W ( f ) × {( A2 ) 2 + ( A3 ) 2 + . . . + ( An ) 2 + ( Anoise ) 2 }
= 100
A1
Equation J. 10
Where
An = amplitude of nth product
Anoise = amplitude of wideband noise and nonharmonic products
W(f) = amplitude weighting function
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J.5
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Continuous spectrum distortion
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. 11). 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 )
=
noncoherent power (300 − 3300Hz )
coherent power (300 − 3300 Hz )
Equation J. 11
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 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. 12 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:
noncoheren t power
coherent power
=
(1 − δ 2 ) Gbb
δ 2 × Gbb
=
(1 − δ 2 )
δ2
=
Gaa × Gbb − Gab
Gab
2
2
Equation J. 12
where
Coherence δ
2
=
Gab
2
Gaa × Gbb
Gaa
= input autospectrum
Gbb
= output autospectrum
Gab
= cross spectrum
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Annex K
(normative)
Send Signal-to-Noise Ratio
K.1 Send signal-to-noise ratio
Send signal-to-noise ratio, SendSNR(f), is a measure of the desired speech transmission relative to unwanted noise
in the room where the phone, handset or headset is used. The measurement is intended to apply to both passive and
active systems. SendSNR(f) is given by equation Equation 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 results are sensitive to the relative levels of the both signals, and may be sensitive to the absolute levels and
types of signals used.

 (G SETP( S + N ) ( f ) − GSETP ( N ) ( f ) )


10
SendSNR ( f ) = 10 log 10
− 1




in dBV / Pa
Equation K. 1
where:
GSETP(S+N)(f) is the RMS power spectrum at SETP with both the mouth simulator and noise sources active
GSETP(N)(f) is the RMS power spectrum at SETP with only the noise source active. The mouth simulator
present, but inactive.
K.2 Weighted send signal-to-noise ratio
Weighted send signal-to-noise ratio, SendSNRw, is a single number which results from applying an intelligibility
weighting WSNR (Table K. 1) to the SendSNR (Equation K. 2).
SendSNRW
=
f = 5000
∑ SendSNR( f ) × W
SNR
f = 200
Equation K. 2
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1/3 octave band
center frequency
200
250
315
400
500
630
800
1000
1250
1600
2000
2500
3150
4000
5000
Weighting
WSNR
.012
.030
.030
.042
.042
.060
.060
.072
.090
.112
.114
.102
.102
.072
.060
Table K. 1 Intelligibility weightings
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Annex L
(normative)
Delay
L.1 General
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 delay range of the measuring equipment
must exceed the expected sidetone delay, or time domain aliasing may occur. The method used should be stated
with the measurement.
L.2 Captured pulse method
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.
L.3 Two-channel analyzer methods
L.3.1
Impulse response method
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.
L.3.2
Cross-correlation method
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.
L.4 Time Delay Spectrometry Method
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.
L.5 MLS Method
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.
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Annex M
(normative)
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).
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 6 or
Equation 7 in 7.6.1. The exact formula depends on the method chosen.
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Annex N
(informative)
Maximum Acoustic Pressure Limits
N.1 ABSTRACT
Both North American and European acoustic pressure limits for telephone headsets are being challenged. Two new
limits at ERP (Ear Reference Point) and DRP (Eardrum Reference Point) are proposed. The new limits are based on
the generally accepted 85dBA 8-hour TWA (Time-Weighted-Average) free-field exposure limit. The TWA allows
the exposure limit to increase 3dB for each time the exposure duration is cut in half, e.g. 88dBA for 4 hours, 91dBA
for 2 hours and so on and so forth. With a 2 second duration (as specified in ITU-T P.360) the allowable free-field
exposure level is 127dBA. Subtract 4dB from 127dBA 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 123dBA. 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 123dBA free-field limit across the frequency bandwidth. The
proposed limits also suggest adding the A-weighting coefficients to simplify actual tests.
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 disputes over the proper telephone headset
acoustic pressure limit on both sides of Atlantic Ocean.
N.2 INTRODUCTION
The two most common telephone headset acoustic pressure limits are the North American frequency dependent
curves at ERP and DRP and the European 118dBA flat (independent of frequency) at ERP.
The North American limit curves were based on United States OSHA (Occupational Safety and Health
Administration) 90dBA 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. 95dBA for 4 hours, 100dBA for 2 hours and so on and so
forth. With a 15-minute duration the allowable free-field exposure level is 115dBA. OSHA regulates the maximum
free-field exposure limit at 115dBA.
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 115dBA 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. 1 and Figure N. 2.
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Bell Lab's T e lephone Headset Acoustic
Pressure Lim it at ER P
140
135
dBSPL
130
125
120
115
110
105
100
100
1000
10000
Frequency (Hz)
Figure N. 1
Bell Lab's T e lephone Headset Acoustic
dBSPL
Pressure Lim it at D R P
140
135
130
125
120
115
110
105
100
100
1000
Frequency (Hz)
Figure N. 2
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ITU-T P.360 explains the derivation of the European 118dBA limit. It was based on an 85dBA 8-hour TWA freefield noise exposure limit. It is 5dB lower than OSHA’s 90dBA limit. The allowable limit increases 3dB for every
halving of exposure duration (Note that OSHA uses a 5dB increment). For the 2-second duration, the allowable limit
is 127dBA. P.360 subtracts 10dB from the 127dBA limit because of “non-occupational exposure”. It subtracts
another 4dB to compensate for narrower telephony bandwidth compared to the free-field broad frequency
bandwidth. P.360 gives 5dB credit to the limit for “sound field” difference (the difference between ERP and freefield). Thus, 127-10-4+5=118dBA. It makes one wonder, which came first: the 118dBA flat limit or its derivation?
The limit is shown in Figure N. 3.
Current Euro Telephone Headset
Acoustic Pressure Limit at ERP
140
135
130
dBA
125
120
115
110
105
100
100
1000
10000
Frequency (Hz)
Current European Limit
Figure N. 3
Both the North American and European limits have their strengths and shortcomings. The North American’s ERP
and DRP limits were based on OSHA’s occupational noise exposure limits. The 90dBA 8-hour TWA free-field limit
has been called too high. Neither the 5dB increment for every halving of duration, nor the absolute maximum of
115dBA, are widely accepted. Nevertheless, American’s method of utilizing frequency dependant transfer functions
to transfer the free-field limit to ERP and DRP limits is correct.
The European limit is based on 85dBA 8-hour TWA free-field limit that is generally accepted. Its increment of 3dB
for every halving of duration is more accepted than OSHA’s 5dB increment. However, transferring free-field limit
to ERP by simply adding 5dB without frequency dependency is hardly justifiable. Subtracting 10dB from the limit
for “non-occupational exposure” is also questionable.
N.3 PROPOSAL
The North American and European limits can be combined.
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Since 85dBA 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 P.360 gives a limit of 127dBA in free-field.
Subtracting 4dB from the 127dBA to compensate the narrower telephony bandwidth, as done in P.360, reduces the
limit to 123dBA. Applying the ERP and DRP transfer functions and adding the A-weighting coefficients to this
123dBA 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. 4 and Figure N. 5.
The 10dB “damage risk” reduction specified in P.360 needs to be re-considered. A reduction to 1/2 of the energy is
3dB, reduction to 1/4 of the energy is 6dB. Figure N. 4 and Figure N. 5 show the preliminary proposed limits at
ERP and DRP and the limits with a 3dB and 6dB reduction for “damage risk”.
Preliminary Proposed Headset
Acoustic Pressure Limits at ERP
140
135
dBSPL
130
125
120
115
110
105
100
100
1000
10000
Frequency (Hz)
w/o Damage Risk
w/ 3dB Damage Risk
w/ 6dB Damage Risk
Figure N. 4
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Preliminary Proposed Headset
Acoustic Pressure Limits at DRP
140
135
dBSPL
130
125
120
115
110
105
100
100
1000
10000
Frequency (Hz)
w/o Damage Risk
w/ 3dB Damage Risk
w/ 6dB Damage Risk
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 the damage risk factor should be further examined and discussed. Hopefully,
this proposal will help in resolving years of disputes over the proper telephone headset acoustic pressure limit on
both sides of Atlantic Ocean.
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Annex O
(normative)
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)
Figure O. 1 Block diagram of simulated speech generator
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Gaussian Signal Generator 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.
X r ( ω ) = cos(2π α − π )
Equation O. 1
X i ( ω ) = sin ( 2 π α − π )
Equation O. 2
where:
α is a random number with uniform distribution 0 ≤ α ≥ 1
The inverse FFT is then taken to transfer the signal to the time domain.
x (n ) ⇔ X i ( ω ) X r (ω )
i
Equation O. 3
The even number segments S(n) are:
Si(n) =Si(n-1)*0.5(1+cos((π (i-0.5)/1024) + Si(n+1)*0.5(1-cos((π (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)
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Annex P
(normative)
ITU-T Recommendation P.50 Noise Bursts Over TDS Sweep
The bias signal consists of random noise with a spectrum and spectrum tolerance conforming to ITU-T
Recommendation P.50 (1993). 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 8,000 Hz, at any rate suitable for Time Delay
Spectrometry (TDS) measurements. The sweeps are not synchronized with the bias pulses. The sweep spectrum is
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. 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).
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Annex Q
(informative)
Use of the Free Field as the Telephonometric Reference Point
[Insert Bob Young’s contribution 00-18 here]
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Annex R
(informative)
Useful Conversion Procedures
R.1 Conversions for dBV to dBm for 600 and 900Ω
Ω
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. 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).
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R.2 Conversions for dBmp to dBrnC for electrical noise measurements
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
R.3 Loudness rating conversions
Conversion from loudness ratings defined in IEEE Standard 661-1979 to those defined in ITU-T Recommendation
P.79 (1993), as specified by TIA/EIA-579A-1998, is as follows:
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 TIA/EIA-579A-1998. Proper conversion may depend upon actual
measurements being made with each measurement standard where frequency responses deviate significantly from
the norm.
Expand to show the frequency responses on which the conversions are based. Get Roger Britt’s data.
R.4 Acoustic sound pressure conventions
dBPa (dB Pascals)
dBSPL (dB Sound Pressure Level)
Where,
0 dBPa = 94 dBSPL, and 0 dBSPL = 20 microPascals, 1 Pa = 1 N/m2
An A weighted sound pressure level in dB (dBSPL, A weighted) is often abbreviated to “dBA”.
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Annex S
(informative)
Loudness Balance Subjective Test Procedure
S.1 Introduction
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.
The methods described in this clause have been successfully used with headsets. In principle, similar methods can
be used with handsets.
S.2 Loudness balance test procedure
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 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
test subjects should have “clinically normal” hearing. That is, the magnitude of measured hearing loss at any test
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
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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.
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.
S.3 Example test circuit
A block diagram of an example test circuit for implementing the loudness balance tests is given in
Figure S. 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.
Amplifier 2
TP 1
Attenuator 1
Source
Signal
Reference
Headset
---Reference
Switch 1
---Test
Amplifier 1
Attenuator 2
TP 2
Attenuator 3
Amplifier 3
Amplifier 4
Test
Headset
Figure S. 1 Loudness Balance Test Circuit
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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.
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.
S.4 Estimate of test headset receive characteristics
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 is the test receiver estimated pressure in dB Pa
RROP is the reference receiver objective pressure in dB Pa
LD
is the loudness difference between test and reference receivers in dB
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