Tansmission Requirements for Wideband Digital Wireline ... Web viewIf codecs or speech frame rates other ... of the measured power referenced to one milliwatt (mW). The ... The words

TIA-PN-920.120-A

(to become TIA-920.120-A)

Telecommunications

Telephone Terminal Equipment

Transmission Requirements for Wideband Digital Wireline Speakerphone

Formulated under the cognizance of TIA Subcommittee TR-41.3Analog and Digital Telephones

With the approval of TIA Engineering Committee TR-41User Premises Telecommunications Requirements

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TIA-PN-920.120-A (to become TIA-920.120-A)

Table of Contents1. SCOPE.........................................................................................................................................1

1.1. LIMITS OF APPLICABILITY.........................................................................................................11.2. LIMITS OF THIS STANDARD FOR PRODUCT EVALUATION.........................................................11.3. CATEGORIES OF CRITERIA.........................................................................................................21.4. ENVIRONMENTAL......................................................................................................................21.5. SAFETY......................................................................................................................................2

2. NORMATIVE REFERENCES.................................................................................................33. DEFINITIONS, ABBREVIATIONS AND ACRONYMS......................................................4

3.1. DEFINITIONS..............................................................................................................................43.2. ABBREVIATIONS AND ACRONYMS.............................................................................................6

4. GENERAL TECHNICAL REQUIREMENTS.......................................................................74.1. TEST ENVIRONMENT – BACKGROUND NOISE...........................................................................74.2. REFERENCE CODECS..................................................................................................................7

4.2.1. L16-256 Reference Codec.............................................................................................84.2.2. Other Reference Codecs.................................................................................................8

4.3. TEST SIGNALS............................................................................................................................84.3.1. Test Stimulus Signal......................................................................................................84.3.2. Analysis..........................................................................................................................9

4.4. PRECAUTIONS............................................................................................................................94.5. TEST ARRANGEMENTS...............................................................................................................9

4.5.1. Reference Codec Arrangement......................................................................................94.5.2. Direct Digital Signal Processing Arrangement............................................................104.5.3. Send General Test Arrangement..................................................................................104.5.4. Receive General Test Arrangement.............................................................................114.5.5. Terminal Coupling Loss General Test Arrangement...................................................11

5. SPEAKERPHONE TECHNICAL REQUIREMENTS........................................................145.1. SEND........................................................................................................................................14

5.1.1. Speakerphone Send Electrical Output..........................................................................145.1.2. Speakerphone Send Frequency Response....................................................................155.1.3. Send Directionality......................................................................................................175.1.4. Speakerphone Send Noise............................................................................................175.1.5. Speakerphone Send Single Frequency Interference.....................................................185.1.6. Speakerphone Send Distortion and Noise....................................................................185.1.7. Speakerphone Send Delay...........................................................................................20

5.2. RECEIVE...................................................................................................................................215.2.1. Speakerphone Receive Acoustic Output......................................................................215.2.2. Speakerphone Receive Volume Control......................................................................225.2.3. Speakerphone Receive Frequency Response...............................................................225.2.4. Speakerphone Receive Noise.......................................................................................245.2.5. Speakerphone Receive Single Frequency Interference................................................255.2.6. Speakerphone Receive Distortion and Noise...............................................................255.2.7. Speakerphone Receive Delay.......................................................................................26

5.3. TALKER SIDETONE CONSIDERATIONS.....................................................................................27

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5.4. WEIGHTED TERMINAL COUPLING LOSS (TCLW)....................................................................28ANNEX A (NORMATIVE) – SPEAKERPHONE RECEIVE LOUDNESS RATING (RLR)

30ANNEX B (NORMATIVE) – SPEAKERPHONE SEND LOUDNESS RATING (SLR)......30ANNEX C (NORMATIVE) – CALCULATION OF LOUDNESS RATINGS.......................31

C.1 WIDEBAND SEND LOUDNESS RATING (HANDSET, SPEAKERPHONE AND HEADSET)..............31C.2 WIDEBAND RECEIVE LOUDNESS RATING (SPEAKERPHONE ONLY).........................................31

ANNEX D (INFORMATIVE) – MEASUREMENT AND LEVEL CONVERSIONS...........33ANNEX E (INFORMATIVE) – R40 PREFERRED FREQUENCIES...................................34ANNEX F (INFORMATIVE) – TERMS RELATED TO DIGITAL TRANSMISSION LEVELS…….......................................................................................................................................37ANNEX G (INFORMATIVE) - RECOMMENDATIONS FOR SUBJECTIVE QUALITY EVALUATION....................................................................................................................................39

G.1 SEND LOUDNESS......................................................................................................................39G.2 SEND NOISE.............................................................................................................................40G.3 SEND DISTORTION....................................................................................................................40G.4 ACOUSTIC ECHO CANCELLATION PERFORMANCE DURING SINGLE-TALK................................41G.5 ACOUSTIC ECHO CANCELLATION PERFORMANCE DURING DOUBLE-TALK...............................42

ANNEX H (INFORMATIVE) – BNR STUDY...........................................................................43H.1 OBJECTIVES.............................................................................................................................43H.2 APPARATUS.............................................................................................................................43H.3 PROCEDURE.............................................................................................................................43H.4 PRELIMINARY RESULTS...........................................................................................................43H.5 RECOMMENDATIONS................................................................................................................44H.6 TABLE OF RESULTS.................................................................................................................44

ANNEX I (INFORMATIVE) – BIBLIOGRAPHY......................................................................46

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Table of FiguresFigure 1 – Digital Telephone Set Test Arrangement with Reference Codec..........................................9Figure 2 – Digital Telephone Set Test Arrangement using Direct Digital Signaling...........................10Figure 3 – Speakerphone Send Test Setup............................................................................................10Figure 4 – Speakerphone Receive Test Setup.......................................................................................11Figure 5 – Terminal Coupling Loss Test Setup....................................................................................11Figure 6 – Speakerphone Send Level Directionality Measurement Setup............................................12Figure 7 – Speakerphone Receive Level Directionality Measurement Setup.......................................13Figure 8 – Speakerphone Send Frequency Response Mask..................................................................16Figure 9 – Speakerphone Receive Frequency Response Mask.............................................................24

Table of TablesTable 1 – Co-ordinates of Speakerphone Send Response Limits.........................................................16Table 2 – Send Directionality Limits....................................................................................................17Table 3 – Limits for Send SDNR vs. Frequency...................................................................................19Table 4 – Limits for Send SDNR vs. Level..........................................................................................19Table 5 – Send SDNR Stimulus Center Frequencies and Band Limits................................................19Table 6 – Send SDNR Analysis Notch Frequency Band......................................................................20Table 7 – Speakerphone Send Delay Requirements.............................................................................21Table 8 – Co-ordinates of Speakerphone Receive Response Limits.....................................................23Table 10 – Limits for Receive SDNR vs. Frequency............................................................................25Table 11 – Limits for Receive SDNR vs. Level....................................................................................25Table 12 – Receive SDNR Stimulus Center Frequencies and Band Limits.........................................26Table 13 – Receive SDNR Analysis Notch Frequency Band...............................................................26Table 14 – Speakerphone Receive Delay Requirements.......................................................................27Table C.1 – Weighting Factors for Calculating Wideband Loudness Ratings and STMR...................32Table E.1 – ISO R40 and R10 Preferred Frequencies, 1/12th and 1/3rd Octave Frequencies................34Table H.1 – Room Noise 35 dBA User Acceptance Levels.................................................................44Table H.2 – Room Noise 45 dBA User Acceptance Levels.................................................................45Table H.3 – Room Noise 55 dBA User Acceptance Levels.................................................................45

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FOREWORD

(This foreword is not part of this standard.)

This document is a TIA Telecommunications standard produced by Working Group TR-41.3.3 of Committee TR-41. This standard was developed in accordance with TIA procedural guidelines, and represents the consensus position of the Working Group and its parent Subcommittee TR-41.3, which served as the formulating group. This standard is based on TIA-920.

There are eight annexes in this Standard. Annex A, Annex B and Annex C are normative and is considered part of this Standard. Annex D, Annex E, Annex F, Annex G, Annex H and Annex I are informative and are not considered part of this Standard.

The leadership of the TR-41.3.3 VoIP and PCM Digital Transmission Performance Working Group (Chair: Allen Woo, Plantronics) acknowledges the written contributions provided by the following individuals in the development of this standard.

Organization Representative Previous Version TIA-920.120-A

Plantronics Allen Woo

Harman Glenn Hess EditorAST Technology Labs Don McKinnon – Editor

Plantronics Steve Graham EditorNortel Networks Roger Britt Editor –

Siemens Ron Magnuson Editor –Alcatel Francois Pinier

AST Technology Labs James Bress –

Cisco Systems Ram Jagadeesan –Cisco Systems Michael Knappe –Cisco Systems Kirit Patel –

IEEE STIT John Bareham

IEEE STIT Rodolfo Ceruti –IEEE STIT Bob Young –

Intel Corporation Mike Dunn –Microsoft Hong Sodoma –

Microsoft Robert Aichner –

Nortel Networks Steve Graham –PictureTel Corporation Dave Lindbergh –

Plantronics Bill Rains –

Polycom Peter Baker –Texas Instruments Warren Karapetian –

Uniden Al Baum –

VTech Steve Whitesell –

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Copyrighted parts of ITU-T Appendix I to Recommendation G.113 and Recommendation P.79 are used with permission of the ITU. The ITU owns the copyright for the ITU Recommendations. Copyrighted parts of ISO 3 are used with permission of the ISO. The ISO owns the copyright for the ISO Standards.

Suggestions for improvement of this standard are welcome. They should be sent to:

Telecommunications Industry AssociationEngineering Department

Suite 300250 Wilson BoulevardArlington, VA 22201

( http://www.tiaonline.org )

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http://www.tiaonline.org/


Introduction

This revision of TIA-920 establishes speakerphone audio performance requirements for wideband digital wireline telephones regardless of protocol or digital format. Significant improvements and corrections have been made. For devices that also have a handset or headset or both, standards TIA-920.110-A and TIA-920.130-A, respectively, apply.

A significant change to this standard is the addition of output level measurements for send and receive testing. Send and receive parameters may be measured using either the output level setting or the loudness rating setting. It is the intent to remove loudness ratings from future versions of this standard.

Wideband is nominally defined as the frequency range between 150 and 6800 Hz. Performance requirements for conventional narrowband telephony in the frequency range between 300 and 3400 Hz are established in ANSI/TIA-810-B.

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Telephone Terminal Equipment – Transmission Requirements for Wideband Digital Wireline Speakerphone Telephones

1. ScopeThis standard establishes transmission performance requirements for speakerphone devices that function as wideband digital wireline telephones. Group speakerphones that are commonly used in large conference rooms and with larger groups of people are outside the scope of this document. Transmission may be over any digital interface including Local or Wide Area Networks, Firewire/IEEE Std 1394, Universal Serial Bus (USB), public ISDN or digital over twisted pair wire. This includes TDM-based and packet-based (e.g. VoIP) transmission protocols. These telephones may be connected through modems, voice gateways, wireless access points, or PBXs, or they may be personal-computer-based telephones. Examples include, but are not limited to: ISDN telephones, digital proprietary telephones, VoIP telephones (corded and cordless), softphones running on personal computers, IEEE Std 802.11 telephones, USB telephony devices, DECT (CAT-iq) telephones, and Bluetooth® telephony devices.

The test measurement methods in this standard reference procedures in IEEE Std 269, IEEE Std 1329 and ITU-T Recommendations. Several performance measurement procedures are established, each of which yields standardized measurement data that may be used for the determination of compliance with this standard. Although this document may reference specific procedures or test equipment, the intent is not to be all-inclusive. Any measurement procedure and equipment that can result in an identical measurement is considered valid.

NOTE: If the main purpose for testing to this standard is comparison testing of different products, rather than compliance testing, then it is important that identical test procedures and equipment be used when testing the different products.

While the procedures may call out specific test points within the requirements, the full range of the requirements take precedence.

1.1. Limits of ApplicabilityThis standard is not intended to describe specific requirements for the following types of digital voice terminal equipment: telephones with carbon transmitters, ISDN terminal adapters, or cellular voice terminals (e.g. cell phones).

1.2. Limits of this Standard for Product EvaluationIt is recognized that there are many parameters that may influence the performance of speakerphone devices including:

1. The distance between the user and the speakerphone device2. Users with a variety of talker levels3. The level and type of environmental noise4. Non-linear processing included in the speakerphone device design (e.g., AGC, AEC)

This standard establishes the minimum performance requirements for speakerphone performance with the following limitations:

1. A single test position is used to simulate the distance between the user and speakerphone.2. One talker level is specified for send tests.3. The speakerphone is tested without environmental noise present during testing.

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4. Testing is performed without consideration for any effects of non-linear processing, although such non-linear processing may be present.

Annex G is included with recommended methods for performing subjective testing which may provide additional information regarding a speakerphone device’s performance. However, this subjective test shall not be a substitution of the objective test nor a condition to pass the standard.

1.3. Categories of CriteriaMandatory requirements are designated by the word "shall". Advisory requirements are designated by the word "should”, or "may”, or "desirable", which are used interchangeably in this standard. Advisory criteria represent product goals or are included in an effort to ensure universal product compatibility. Where both a mandatory and an advisory level are specified for the same criterion, the advisory level represents a goal currently identifiable as having distinct compatibility or performance advantages toward which future designs should strive.

1.4. EnvironmentalThis standard does not contain environmental requirements. Environmental requirements can be found in ANSI/TIA-571-B.

1.5. SafetyThis standard does not contain safety requirements. Compliance with the applicable UL and CSA safety standards may be required in certain locations.

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2. Normative ReferencesThe following standards contain provisions, which, through reference in this text, constitute provisions of this Standard. At the time of publication, the editions indicated were valid. All standards are subject to revision, and parties to agreements based on this Standard are encouraged to investigate the possibility of applying the most recent editions of the standards indicated below. ANSI and TIA maintain registers of currently valid national standards published by them.

1. ANSI S1.4-1983 (R2006)/ANSI S1.4a-1985 (R2006), Sound Level Meters.2. IEEE Standard 269-2010, Standard Methods for Measuring Transmission Performance of

Analog and Digital Telephone Sets, Handsets, and Headsets.3. IEEE Standard 1329-2010, Standard Method for Measuring Transmission Performance of

Speakerphones.4. ITU-T Recommendation G.100.1 (2001-11), The use of decibel and of relative levels in

speechband telecommunications.5. ITU-T Recommendation G.122 (1993-03), Influence of national systems on stability and talker

echo in international connections.6. ITU-T Recommendation G.722 (1988-11), 7 kHz audio-coding within 64 kbit/s.7. ITU-T Recommendation P.79 (2007-11), Calculation of loudness ratings for telephone sets.8. ITU-T Recommendation P.501 (2009-12), Test signals for use in telephonometry.

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3. Definitions, Abbreviations and AcronymsFor the purposes of this standard, the following definitions apply.

3.1. DefinitionsActive Speech Level:The active level of speech or speech like signals using a speech voltmeter as defined in ITU-T P.56, method B. Measures the active speech level by accounting for pauses that are a natural part of a speech signal.

Artificial Ear/Mouth and Ear/Mouth Simulators:This standard uses the terms “ear simulator” and “mouth simulator” synonymously with the terms “artificial ear” (ITU-T P.57) and “artificial mouth” (ITU-T P.51), respectively, to harmonize with IEEE Std 269.

Codec:A codec is a combination of an analog-to-digital encoder and a digital-to-analog decoder operating in opposite directions of transmission in the same equipment. An example of such a codec is the reference codec, which is described later. In this document if the context does not indicate the reference codec, then the term codec refers to the digital-to-digital voice signal coder and encoder.

Cordless:A cordless terminal does not have a cable between its handset, headset or speakerphone transducers and a base unit that connects to the network.

Direct Digital Signal Processing:Direct digital signal processing utilized digital means of signal generation and analysis and avoids the use of an analog network interface.

Electric Power and Noise Levels:The following electric power and noise level units are used in this standard:

The rms level of a sine wave with a peak value which is a specified number of dB below digital overload is defined to be 0 dBm0. This shall be considered equivalent to 0 dBm (0.775 Vrms for 600 ohms) in an analog system. Each different codec will have its own specified 0 dBm0 level relative to digital overload. See ITU-T G.100.1.

NOTE: 0 dBm0 is generally not the maximum digital code.

Far-end:The term far-end is used to describe the part of the end-to-end connection opposite where the device under test (DUT) is located.

Loudness Ratings:The loudness rating algorithm is defined in ITU-T Recommendation P.79. Loudness ratings are calculated from a telephone’s frequency response measurement data. It is a single number metric that describes the loudness loss of the connection. The louder the telephone the less positive is of the loudness rating.

Mouth Reference Point (MRP):The mouth reference point is located on axis and 25 mm in front of the lip plane of a mouth simulator.

Near-end:The term near-end is used to describe the part of the end-to-end connection where the DUT is located.

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Nominal Volume Control Setting:The receive volume control setting that results in the closest measurement to the nominal RLR ‘Nom VC RLR’ or nominal Level value ‘Nom VC Level’.

Nom VC Level:The receive volume control setting that results in the closest measurement to the nominal receive level requirement.

Nom VC RLR:The receive volume control setting that results in the closest measurement to the nominal RLR requirement.

Preferred Free Field Microphone (FF Microphone):The preferred receive measurement device is a free field microphone located at the standard test position, or the manufacturers recommended listening position. For alternative measurement devices such as ear simulators associated with using a head and torso simulator (HATS), see relevant sections of IEEE Std 269 and IEEE Std 1329.

Recommended Test Position (RTP):A Recommended Test Position (RTP) may be defined by the manufacturer for use in testing a speakerphone instead of the Standard Test Position. See IEEE Std 1329 for information regarding defining an RTP.

Reference Codec:A codec that approaches an ideal codec2 and has superior, well-defined, characteristics. Although power levels are referenced to 600 ohms, the reference codec does not require a physical 600 ohm source resistor.

Sound Pressure Level:Sound pressure level is a value expressed as a ratio of the pressure of a sound to a reference pressure. The following sound level units are used in this standard:

dBPa: 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 this sound to the reference pressure of 1 Pascal (Pa).NOTE: 1 Pa = 1 N/m2

dBSPL: 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 this sound to the reference pressure of 2 X 10-5 N/m2.Where: 0 dBSPL = 20 micro-Pascals and 0 dBPa = 94 dBSPL.

dBA: The A-weighted sound level is the sound pressure level in dBSPL, weighted by use of metering characteristics and A-weighting specified in ANSI S1.4.

2 Ideal codec characteristics, such as attenuation versus frequency distortion, idle channel noise, and quantizing distortion are specified in ITU-T Recommendation G.712.

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There is no need for this paragraph. It repeats the definitions of "Nom VC Level" and "Nom VC RLR" below. We may want to consider to make the Nom VC Level and Nom VC RLR as two bullets points of "Nominal Volume Control Setting:"


Change to Codec2


3.2. Abbreviations and AcronymsAbbreviations and acronyms, other than in common usage, which appear in this standard, are defined below.

AEC Acoustic Echo CancellerAGC Automatic Gain ControlASL Active Speech LevelCAT-iq Cordless Advanced Technology-internet and qualityCCW Counter Clock WiseCPE Customer Premises EquipmentCSS Composite Source SignalDECT Digital Enhanced Cordless TelecommunicationsERP Ear Reference PointFFT Fast Fourier TransformFF Free FieldHATS Head and Torso SimulatorISDN Integrated Services Digital NetworkL16-256 Linear, Sixteen Bit, 256 kbit/s CodecMRP Mouth Reference PointOLR Overall Loudness RatingPBX Private Branch ExchangePCM Pulse Code ModulationRLR Receive Loudness RatingRFC Request for Comment (used by the IETF to define IP Protocol)RTP Recommended Test PositionSDNR Signal to Distortion and Noise RatioSLR Send Loudness RatingSTMR Sidetone Masking RatingSTP Standard Test PositionTCLT Temporally weighted Terminal Coupling LossTCLw Weighted Terminal Coupling LossTDM Time Division MultiplexingVAD Voice Activity DetectorVC Volume ControlVoIP Voice over Internet Protocol

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4. General Technical Requirements

4.1. Test Environment – Background NoiseThe test environment requirements for speakerphone testing are defined in IEEE Std 1329 with one specific requirement of having the background noise level not exceeding 29 dBA. Several of the tests in this standard require a quiet environment. If the measurement level and background noise level ratio is less than 10 dB the measurement is not considered to be valid.1. Example one

Background noise = 28 dBA (Lroom) Receive noise measurement = 34.5 dBA (Lnoise) (failed) Lnoise – Lroom = 6.5 dB This would not be considered a valid measurement.

Per IEEE Std 1329, if the background noise level cannot be further reduced the test microphone may be repositioned at half the standard test position distance along the 50 cm axis shown in 4.5.4. With the measurement repeated:2. Example two

Background noise = 28 dBA (Lroom) (no change) Receive noise measurement = 40.5 dBA (½Lnoise) ½Lnoise – Lroom = 12.5 dB This would be considered a valid measurement. For reporting, the measurement would be

referenced back to the STP by subtracting 6 dB from the measured value.

It should be noted that the low level input tests for SDNR may also be impacted by background noise.3. Example three

Background noise = 28 dBA (Lroom) SDNR fundamental measurement = 58 dBSPL (Lfund) Notched SDNR noise measurement = 33 dBA (Lmeasure) Calculated SNR = 25 dB (failed) Lmeasure – Lroom = 5 dB This would be not considered a valid measurement due to the noise ratio.

4.2. Reference CodecsThis standard describes the use of an L16-256 reference codec for testing purposes. If the telephone supports the L16-256 codec, then the L16-256 reference codec shall be used when testing for compliance with this standard. If the L16-256 codec is not supported, then a telephone is considered compliant to this standard if all requirements of this standard are met when tested with at least one of the telephone's supported codecs.

NOTES:1. For telephones where single or tandem codecs other than L16-256 are used (e.g. cordless

interface to a telephone set), the codec may affect the test results and some voice transmission technologies may be unable to meet specified noise and/or distortion requirements. Such devices need further investigation.

2. Telephones using non-linear voice signal processing may require subjective testing.

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Change this sentence to: If the difference between the measurement leve and the background noise level is less than 10 dB the measurement is not considered to be valid.


4.2.1. L16-256 Reference CodecThe L16-256 codec is specified as follows:

L16-256 uses uncompressed 16-bit linear PCM coding sampled at 16 kHz and having a bit rate of 256 kbit/s. The L16-256 coder shall use 16-bit signed representation with 65535 equally divided steps between minimum and maximum signal level from -32768 to 32767.

Each L16-256 sample value will be represented in two’s complement notation and transmitted in network byte order with the most significant byte first.

The L16-256 0 dBm0 level is 3.17 dB below the digital overload point. This means that a L16-256 coded sine wave that has an rms level of 0 dBm0 has peak values which are 3.17 dB below digital full scale.

The L16-256 quiet code is the digital code representing the smallest encoded analog level. Full scale code is the digital code representing the largest encoded analog level.

A 1 kHz 0 dBm0 sine wave can be represented by the following values:Sample [ 0] = 0Sample [ 1] = 8705Sample [ 2] = 16085Sample [ 3] = 21016Sample [ 4] = 22748Sample [ 5] = 21016Sample [ 6] = 16085Sample [ 7] = 8705Sample [ 8] = 0Sample [ 9] = -8705Sample [10] = -16085Sample [11] = -21016Sample [12] = -22748Sample [13] = -21016Sample [14] = -16085Sample [15] = -8705

4.2.2. Other Reference Codecs The performance requirements in this standard were developed based on testing with an L16-256 reference codec. The performance requirements and testing methods specified in this standard may also be applicable for devices using codecs other than L16-256. If a codec other than L16-256 is used, the 0 dBm0 level of the codec, and a means to interface to and from this codec should be provided by the device manufacturer or a relevant standard. (Example: For a G.722 codec, 0 dBm0 = 9 dB below full scale.) If the 0 dBm0 level is not provided, then 3.17 dB below full scale may be assumed. (Example: In most pc-based applications in which no other definition supersedes, use 3.17 dB.)

NOTE: 0 dBm0 level has a significant effect on measurements. It is important to state the level being used for the test.

4.3. Test Signals

4.3.1. Test Stimulus SignalUnless stated otherwise, the test stimulus signal shall be the uncompressed male speech signal published in IEEE Std 269.

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The bandwidth of the test signal used shall nominally cover 100 Hz to 8000 Hz unless otherwise specified. The test signal levels specified in this standard shall be used.

The level of the test signal shall be measured using ITU-T P.56, method B, which measures the active speech level by accounting for pauses that are a natural part of the speech signal.

NOTES:1. Algorithmic processes, such as Echo Control, VAD and AGC, may influence the test results.2. If ASL is not used the measured level will be 0.3 dB lower.

4.3.2. AnalysisAnalysis shall be done in 1/12th octave bands unless otherwise specified.

4.4. PrecautionsCoding, decoding, packetization and other signal processing may introduce significant delays that must be accounted for by the measurement system. Refer to IEEE Std 269 for additional precautions regarding test signal usage.

Telephones using non-linear voice signal processing may require subjective testing to validate or supplement objective measurement.

4.5. Test Arrangements

4.5.1. Reference Codec ArrangementA reference codec, as shown in Figure 1, may be used for testing digital telephone terminals with analog test equipment. The interface block passes the voice channel digital bit stream to the terminal without modification. There is no gain or loss in the receive direction due to the interface. If the interface does change the digital voice stream, then the terminal and interface shall be considered jointly as the terminal. An example of this is a receive volume control implemented in a PBX or gateway.

Figure 1 – Digital Telephone Set Test Arrangement with Reference Codec

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4.5.2. Direct Digital Signal Processing ArrangementDirect digital signal generation of the receive signal and analysis of the send signal may be used as shown in Figure 2. This arrangement may be used in place of the reference codec. The interface block passes the voice channel digital bit stream to the terminal without modification. There is no gain or loss in the receive direction due to the interface. If the interface does change the digital voice stream then the terminal and interface shall be considered jointly as the terminal.

Figure 2 – Digital Telephone Set Test Arrangement using Direct Digital Signaling

4.5.3. Send General Test ArrangementThe 50cm reference test position and send general test setup with the mouth simulator is shown below in Figure 3.

Figure 3 – Speakerphone Send Test Setup

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4.5.4. Receive General Test ArrangementThe 50cm reference test position receive general test setup with free field microphone (FF) is shown below in Figure 4.

Figure 4 – Speakerphone Receive Test Setup

4.5.5. Terminal Coupling Loss General Test Arrangement

Figure 5 – Terminal Coupling Loss Test Setup

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CPE

300DegreePosition

330DegreePosition

60DegreePosition

30DegreePosition

0DegreePosition

Figure 6 – Speakerphone Send Level Directionality Measurement SetupNOTES:1. The Speakerphone setup at the 0 Degree position is using the setup in Figure 3.2. For the other test positions, the Speakerphone is rotated around its physical center counter

clockwise.3. Test table is at least 1 meter square.

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CPE

300DegreePosition

330DegreePosition

60DegreePosition

30DegreePosition

0DegreePosition

Figure 7 – Speakerphone Receive Level Directionality Measurement SetupNOTES:1. The Speakerphone setup at the 0 degree position is using the setup in Figure 4.2. For the other test positions, the Speakerphone is rotated around its physical center.3. The test table is at least 1 meter square.

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Figure 7 should be deleted as it is not used anywhere in the document.


5. Speakerphone Technical RequirementsThe requirements of this section apply to a stand-alone speakerphone or speakerphone interface device and terminal together.

All measurements are performed using the Standard Test Positions (STP) shown in Figure 3 and Figure 4, as described in IEEE Std 1329, or at the manufacturers Recommended Test Position (RTP). However the device is tested, the test position used shall be documented for all tests.

Algorithmic processes, such as Echo Control, VAD and AGC, may influence the test results and necessitate the need for preconditioning before the measurement is made.

All tests shall be performed with the receive volume control setting that provides an output level or RLR closest to 65 dBSPL (Nom VC Level), or +16 dB (Nom VC RLR) respectively as determined in clause 5.2.1.

If the device has an adjustable send (microphone) level control, the setting that provides an output level or SLR closest to -25 dBm0, or +13 dB respectively shall be determined per the measurement method in section 5.1.1 and used for all applicable send testing.

5.1. SendThe send performance of a telephone is determined by its acoustical-to-electrical transfer characteristics. To determine the send performance, requirements for frequency response, Send output level, Send Loudness Rating (SLR), distortion, noise and delay are specified.

The send parameters shall be measured using the device settings that comply with either the output level settings or loudness rating settings. The send loudness rating clauses are included in normative (Normative) – Speakerphone Receive Loudness Rating (RLR) Annex B.

5.1.1. Speakerphone Send Electrical Output To ensure that speakerphone send path provides an optimal output signal with a normal speech levels.

5.1.1.1. RequirementWith a -4.7 dBPa input signal level (nominal), the device shall have either:1. A send output level of -25 dBm0, with a tolerance of ±4.0 dB.2. A send loudness rating (SLR) level of +13 dB, with a tolerance of ±4.0 dB.

NOTES:1. See Annex B for more information regarding SLR.2. This output level and SLR target is 5 dB quieter relative to a handset due to higher talker

levels, small frequency response differences and a desire to reduce unwanted ambient noise to be transmitted to the far-end listener.

3. The nominal test position angle for a device is the direction that a single user is most likely to face. Typically, this is the STP and would be the 0 degree position.

4. If the device has an adjustable send (microphone) level control, the level control may need to be adjusted to meet this requirement.

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delete "a"


The title of Annex B is "Speakerphone Send Loudness Rating (SLR).Shouldn't the receive VC set to the nominal level?


5. The device and/or system may have Send Automatic Gain Control (AGC), that may alter the output level or SLR depending on incoming levels. The AGC should not affect the output level or SLR when the STP and nominal test position angle is used, and test speech levels are −4.7 dBPa.

5.1.1.2. Measurement Method1. Level: Using the STP of Figure 3, apply the -4.7 dBPa input signal level and measure the ASL of the

output signal.2. SLR: Use the 1/3rd octave sensitivity data collected from the send frequency response measurement

in 5.1.2 to calculate SLR using Equation C-1 of Annex C and bands 1 to 20 of Table C.1.

5.1.2. Speakerphone Send Frequency ResponseThe send frequency response is the ratio of the voltage output of the reference codec, or digital bit stream equivalent, to the sound pressure at the Mouth Reference Point (MRP) for each frequency or frequency band (Fi) as shown in Equation 1 below:Equation 1:

SMJ = 20 log (VSEND / PM) dB rel 1 V / PaWhere

SMJ Send Sensitivity, Mouth to Junction, at Fi.PM Sound pressure at the MRP at Fi.VSEND RMS output voltage of the reference codec, or digital bit stream equivalent, at F i.

Because frequency response calculation incorporates division, pauses in the speech signal shall be handled the same for the stimulus and measurement.

5.1.2.1. RequirementThe send frequency response shall be within the upper limit and the lower limits given in Table 1 and shown in Figure 8 for electrical output measurements based on output level or loudness rating.

The frequency response is plotted using the 1/3rd octave band center frequency measurement points on a linear dB scale against frequency on a logarithmic scale. The frequency response mask is a floating or “best fit” mask.

5.1.2.2. Measurement MethodThe send frequency response is measured using the STP of Figure 3, with a test signal level of -4.7 dBPa. Measure the frequency response in 1/3rd octave bands, over a range of 100 Hz through 8000 Hz.

22

581582583584585586587588589590591592593594595596597598599600601602603604605606607608609610611612613614615616


add "at the MRP"


The ballot in PDF has an error message


Table 1 – Co-ordinates of Speakerphone Send Response Limits

Nominal1/3 Octave Band

(Hz)

Lower1/3 Octave Band Edge

(Hz)

Upper1/3 Octave Band Edge

(Hz)

Send Response

Limit(dB)

Upper Limit100 to 200 90 224 2.0250 to 1000 224 1120 4.0

1250 to 1600 1120 1800 5.02000 to 2500 1800 2800 6.03150 to 4000 2800 4500 7.0

5000 4500 5600 6.06300 to 8000 5600 9000 4.0

Lower Limit250 224 280 -8.0315 280 355 -6.0

400 to 3150 355 3550 -4.04000 3550 4500 -6.05000 4500 5600 -8.0

-15

-10

-5

0

5

10

100 1000 10000

Arb

itrar

y Le

vel (

dB)

Frequency (Hz)

Send Frequency Response Template

Figure 8 – Speakerphone Send Frequency Response Mask

2324

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618

619620621


5.1.3. Send Directionality To verify electrical output stability when stimulus is applied an angle off center to the normal test position.

5.1.3.1. RequirementThe send directionality values relative to the 0 degree position shall be within the limits given in Table 2 for electrical output measurements based on output level or loudness rating.

Table 2 – Send Directionality LimitsAngle

Degrees (CCW)Level Directionality

(dB)Loudness Rating Directionality

(dB)

30 +/- 3 +/- 360 < 3 < 3

300 < 3 < 3330 +/- 3 +/- 3

NOTES:1. The intent is to have consistent output when the talker is +/- 30 degrees from the center axis.2. The intent is to ensure that there is no more than 3 dB gain when the talker is +/- 60 degrees

from the center axis.

5.1.3.2. Measurement MethodThe speakerphone is rotated about its physical center counter-clockwise taking care to keep the MRP of the artificial mouth at a constant distance from the physical center of the speakerphone, see Figure 6. The send output is first measured at the 0 degree angle according to 5.1.1, then rotate the speakerphone about its physical center to the 30, 60, 300, and 330 degree positions repeating the output measurement at each position. Calculate the directionality difference for each test angle and the output at 0 degrees.

5.1.4. Speakerphone Send NoiseThe send noise of a digital telephone is the 5 second average noise level at the digital transmit output with the speakerphone microphone isolated from sound input and mechanical disturbances.

5.1.4.1. RequirementThe overall send noise shall be less than or equal to -63 dBm0, A-weighted for electrical output measurements based on output level or loudness rating.

NOTE: The speakerphone microphone will not be as close to the MRP as a handset microphone, thus requiring more gain, and it will be subject to masking from background room noise.

5.1.4.2. Measurement MethodPlace the device in a quiet environment using the STP shown in Figure 3. Apply the male speech stimulus signal and begin the measurement 500 ms (± 50 ms) after the end of the active part of the stimulus signal. Measure the A-weighted, 5 second average, noise level at the digital interface output or the reference codec decoder output over the frequency range of 100 to 8000 Hz.

24

622623624625626627628629

630631632633634635636637638639640641642643644645646647648649650651652653654655656657


See above in [A11]


"<3" should be changed to "<6", otherwise this makes no sense.


It doesn't make sense that it's okay to be -3dB at 30 and 330 degrees but not okay at 60 and 300 degrees. Replace the table with one sentance reads "no more than ±3 dB within 60 degrees off center".


NOTE: A-weighting is specified for this measurement because traditional Psophometric weighting for electrical measurements assumes a narrowband receiver at the far-end phone.

5.1.5. Speakerphone Send Single Frequency InterferenceNarrow-band noise, including single frequency interference, 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 receive noise level.

5.1.5.1. RequirementThe A-weighted send single frequency interference shall be less than -70 dBm0 for electrical output measurements based on output level or loudness rating.

5.1.5.2. Measurement MethodUsing the measurement method in clause 5.1.4, measure the A-weighted noise level at the digital interface output or the reference codec decoder output with an effective bandwidth of not more than 31 Hz, over the frequency range of 100 to 8000 Hz. If FFT analysis is used, then “Flat Top” windowing is employed. The peak level between 100 Hz and 8000 Hz is the single frequency interference value.

5.1.6. Speakerphone Send Distortion and NoiseThe send distortion and noise is measured using pulsed 1/3 octave noise.

5.1.6.1. RequirementFor electrical output measurements based on output level or loudness rating:1. The signal power to the total distortion and noise power ratio (SDNR) shall be above the limits given

in Table 3 and Table 4.2. It is recommended that the send distortion and noise performance also be confirmed using subjective

listening tests.

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Table 3 – Limits for Send SDNR vs. FrequencyCenter Frequency

(Hz)SDNR (dB)

(+5 dBPa Send Level)315 24400 26800 26

1600 262500 26

Table 4 – Limits for Send SDNR vs. LevelSend Level at the MRP

(dBPa)SDNR (dB)

(800 Hz)-20 22-15 26-10 26-5 260 26

NOTES:1. 26 dB = 5%2. An additional test frequency at 315 Hz has been added since acoustic energy levels may be

much higher at this frequency on a speakerphone than with a handset and headset, especially when considering gradient response microphones and the echo path.

5.1.6.2. Measurement Method1. Send distortion and noise is measured according to IEEE Std 269 using the setup shown in Figure 3.2. The test stimulus is white noise band limited to 1/3 rd octave and pulsed 250 ms ON (active period),

150 ms OFF (conditioning period). This sequence should be repeated ten times for a total stimulus of four seconds. This test signal should place the telephone in a well-defined, reproducible state for the period of the measurement.

3. If the device does not maintain the reproducible state for the entire measurement period, then the 150 ms silent conditioning period may be replaced with an alternative conditioning signal (e.g. real speech signal, longer “OFF” period, etc.).

4. Apply the test signal at the levels given in Table 3 and Table 4 using the center frequencies and band limits specified in Table 5.

Table 5 – Send SDNR Stimulus Center Frequencies and Band LimitsCenter Frequency

(Hz)Band Edges

(Hz)

315 280 355400 320 480800 675 9251600 1375 18152500 2205 2855

26

687

688

689690691692693694695696697698699700701702703704705706707

708


5. Process the last 200 ms of each measured noise burst using Hanning windowing with 5 Hz resolution FFT from 100 Hz to 8000 Hz. After time averaging, remove the data within the notch frequency band specified in Table 6.

Table 6 – Send SDNR Analysis Notch Frequency BandCenter Frequency

(Hz)Notch Band Edges

(Hz)

315 275 360400 315 485800 670 9301600 1370 18202500 2200 2860

6. Calculate the ratio of the signal power to the total A-weighted distortion and noise power of the signal output.

5.1.7. Speakerphone Send DelayDelay is a complex end-to-end issue. Certain aspects of delay can be optimized, such as the internal hardware/firmware delay and the optimization of the jitter buffer operation, which must trade-off the impairment of packet loss against the expected delay variation of the far-end telephone and/or the network. Other aspects, such as packetization and depacketization are also important sources of delay but they are a function of the selected codec and the number of speech frames per packet, so they cannot be optimized.

When reporting compliance with this standard, only the category with the largest measured delay shall be reported if the send and receive categories are different. If codecs or speech frame rates other than those specified in the measurement methods are used, then they must be clearly identified when reporting compliance.

The send delay is defined here as the time from when an acoustic signal leaves the mouth simulator playing into a device to the time its digitized representation arrives at the digital network interface.

5.1.7.1. RequirementFor electrical output measurements based on output level or loudness rating:1. Corded speakerphone terminals shall be configurable so that requirements of at least Category B are

met.2. Cordless speakerphone terminals should be configurable so that the requirements of at least Category

C are met.3. The compliance category of the product shall be the same as the longest measured latency category

for send or receive. Products that do not meet category C cannot be categorized.

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714715716717718719720721722723724725726727728729730731732733734735736737738739740741742743744745


Table 7 – Speakerphone Send Delay Requirements

Category Delay Time (ms)

A ≤ 35B ≤ 55C ≤ 70

NOTE: This standard specifies delay in terms of categories for network planning purposes, similar to ITU-T Recommendation P.1010.

5.1.7.2. Method of MeasurementA digital audio measuring device capable of measuring the delay between an injected signal (to the mouth simulator) and a digitally transmitted signal is connected to the mouth simulator and directly to the network output of the telephone. All delays inherent in the measurement system itself must be subtracted from the overall delay measurement to obtain the exact Send delay.

If the telephone uses packet based transmission, then the telephone should be set to transmit packets with a maximum speech frame rate of 20 ms and with one speech frame per packet.

NOTE: To minimize the delay measurement uncertainty, it is recommended to use the shortest possible packet size for the delay measurement.

An acoustic signal of -4.7 dBPa shall be generated at the mouth simulator. The delay between the time the pulse left the mouth to the time it was received at the telephone’s packet network interface shall be measured. The send delay shall be used to determine the corresponding category.

5.2. ReceiveThe receive performance of a telephone is determined by its electrical-to-acoustical transfer characteristics. To determine the receive performance, requirements for frequency response, receive output level, Receive Loudness Rating (RLR), receive Sound Pressure Level (dBSPL), distortion, noise, volume control and delay are specified.

The receive parameters shall be measured using the device settings that comply with either the output level settings or loudness rating settings.

All receive tests shall be performed with the receive volume control set to the applicable Nominal Volume Control Setting determined in clause 5.2.1, Nom VC Level, or to the Nom VC RLR setting. The receive loudness rating clauses are included in normative Annex A.

5.2.1. Speakerphone Receive Acoustic Output The receive performance of a telephone is determined by its electrical-to-acoustical transfer characteristics.

5.2.1.1. RequirementWith a -20 dBm0 input signal level (nominal), the device shall have a volume control setting that provides either:1. A receive output level of 65 dBSPL, with a tolerance of ±4.0 dB. If more than one volume control

setting meets this requirement, the setting that produces an output level closest to 65 dBSPL is

28

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add "norminal" before acoustic


Delete "receive output level" as we already have receive Sound Pressure Level (dBSPL)?


defined as the Nom VC Level setting used for other measurements in this standard.2. A receive loudness rating (RLR) level of +16 dB, with a tolerance of ±4.0 dB. If more than one

volume control setting meets this requirement, the setting that produces an output level closest to an RLR value of +16 dB is defined as the Nom VC RLR setting used for other measurements in this standard.

NOTES:

1. See Annex A for more information regarding RLR.

2. The nominal RLR value of +16 dB is calculated using the formula in Annex C and does not include the 14 dB correction factor used in other standards (see IEEE Std 1329 and ITU-T Recommendation P.340) speakerphone/handsfree RLR calculations.

5.2.1.2. Measurement Method1. Level: Using the STP of Figure 4, apply the -20 dBm0 input signal level and measure the ASL of the

output signal.2. RLR: Use the 1/3rd octave sensitivity data collected from the receive frequency response

measurement in 5.2.3 to calculate RLR using Equation C-2 of Annex C and bands 1 to 20 of TableC.1.

5.2.2. Speakerphone Receive Volume Control A speakerphone is expected to receive a wide dynamic range of input speech levels. A speaker volume control (manual or automatic or both) provides a means to compensate for this variable input level. Different volume control settings may be used to meet the requirements at the different test input levels.

5.2.2.1. Requirement1. With a -20 dBm0 input signal level (nominal), the device shall have a volume control setting that will

provide a minimum receive output level of 70 dBSPL.2. With a -35 dBm0 input signal level (quiet), the device shall have a volume control setting that will

provide a minimum receive output level of 65 dBSPL.3. To ensure adequate volume control attenuation, the device shall have a volume control setting that

will produce a receive level of 70 dBSPL or lower with a -10 dBm0 test level (loud).

NOTES:1. To avoid clipping the test signal for the -10 dBm0 input level, use the compressed version of

the male speech signal in IEEE Std 269. 2. The speech levels are defined in IEEE Std 1329.

5.2.2.2. Measurement MethodUsing the STP of Figure 4, apply the -20 dBm0 input signal level and measure the ASL of the output signal. Repeat measurement as required to find the volume setting that meets the -20 dBm0 input level requirement. Repeat for the -35 dBm and -10 dBm0 test input levels.

5.2.3. Speakerphone Receive Frequency ResponseThe receive frequency response is the ratio of the sound pressure measured at the free field measurement microphone located at the Receive Test Point (typically at 50cm) to the voltage input to the reference codec, or digital bit stream equivalent, for each frequency or frequency band (Fi) as shown in Equation 2 below:

2930

787788789790791792793794795796797798799800801802803804805806807808809810811812813814815816817818819820821822823824825826827828829830831


Equation 2:SJE = 20 log (PE / VRCV) dB rel 1 Pa / V

WhereSJE Receive Sensitivity, Junction to free field microphone (Ear), at Fi.PE Sound pressure measured by free field microphone at Fi.VRCV RMS input voltage to the reference codec, or digital bit stream equivalent, at F i.

Because frequency response calculation incorporates division, pauses in the speech signal shall be handled the same for the stimulus and measurement.

5.2.3.1. RequirementThe receive frequency response shall be within the upper and lower limits given in Table 8 and shown in Figure 9.

The frequency response is plotted using the 1/3rd octave band center frequency measurement points on a linear dB scale against frequency on a logarithmic scale. The frequency response mask is a floating or “best fit” mask.

5.2.3.2. Measurement MethodThe receive frequency response is measured using the STP in Figure 4, with a test signal level of -20.0 dBm0. Measure the frequency response in 1/3rd octave bands, over a range of 100 Hz through 8000 Hz.

Table 8 – Co-ordinates of Speakerphone Receive Response Limits

Nominal1/3 Octave Band

(Hz)

Lower1/3 Octave Band Edge

(Hz)

Upper1/3 Octave Band Edge

(Hz)

Receive Response

Limit(dB)

Upper Limit100 to 200 90 224 4.0

250 to 4000 224 4500 5.0

5000 to 8000 4500 9000 4.0

Lower Limit250 224 280 -9.0

315 280 355 -7.0

400 to 2500 355 2800 -5.0

3150 to 4000 2800 4500 -6.0

5000 4500 5600 -9.0

30

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855


-15

-10

-5

0

5

10

100 1000 10000

Arb

itrar

y Le

vel (

dB)

Frequency (Hz)

Receive Frequency Response Template

Figure 9 – Speakerphone Receive Frequency Response Mask

5.2.4. Speakerphone Receive NoiseThe receive noise of a digital telephone is the 5 second average noise level.

5.2.4.1. Requirement1. The overall receive noise shall be less than 34 dBA for the Nominal Volume Control setting.2. The overall receive noise shall be less than 39 dBA for the volume control setting that meet the

requirements as specified in section 5.2.2 for the -35 dBm0 input signal level.

NOTE: Normal room noise is somewhere between 35 dBA to 50 dBA. Less than 34 dBA noise from the speakerphone would mostly be masked by room noise.

5.2.4.2. Measurement MethodA signal corresponding to a decoder quiet code is applied at the digital interface. Using the STP of Figure4, measure the A-weighted, 5 second average, noise level over the frequency range of 100 to 8000 Hz. If the measured noise level is less than 10 dB above the background noise level, the free field microphone may be repositioned at half the standard test position distance along the 50 cm axis (see IEEE Std 269).

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add: "See Clause 4.1 and"


5.2.5. Speakerphone Receive Single Frequency InterferenceNarrow-band noise, including single frequency interference, 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 receive noise level.

5.2.5.1. RequirementThe receive A-weighted single frequency interference shall be 10 dB quieter than the overall A-weighted receive noise measured in clause 5.2.4.

NOTE: For overall A-weighted receive noise levels lower than 25 dBA, single frequency interference requirements and measurements are under study.

5.2.5.2. Measurement MethodUsing the same measurement method used in clause 5.2.4, measure the A-weighted, 5 second average, noise level with an effective bandwidth of not more than 31 Hz, over the frequency range of 100 Hz to 8000 Hz. If FFT analysis is used, then “Flat Top” windowing shall be employed. The peak level between 100 Hz and 8000 Hz is the single frequency interference value.

5.2.6. Speakerphone Receive Distortion and NoiseThe receive distortion and noise is measured using pulsed 1/3 octave noise.

5.2.6.1. Requirement1. The ratio of the signal power to the total A-weighted distortion and noise power (SDNR) should be

greater than or equal to the limits given in Table 9 and Table 10.2. It is recommended that the receive distortion and noise performance also be confirmed using

subjective listening tests.

Table 9 – Limits for Receive SDNR vs. FrequencyCenter Frequency

(Hz)SDNR (dB)

(-10 dBm0 Test Level)

400 26800 26

1600 262500 26

Table 10 – Limits for Receive SDNR vs. LevelReceive Level

(dBm0)SDNR (dB)

(800 Hz)

-35 22-30 26-25 26-20 26-15 26

32

875876877878879880881882883884885886887888889890891892893894895896897898899900901902

903


NOTE: 26 dB = 5%

5.2.6.2. Measurement Method1. Receive distortion and noise is measured using the STP in Figure 4 with the room noise less than 29

dBA.2. The test stimulus is white noise band limited to 1/3 rd octave and pulsed 250 ms ON (active period),

150 ms OFF (conditioning period). This sequence should be repeated ten times for a total stimulus of four seconds. This test signal should place the telephone in a well-defined, reproducible state for the period of the measurement.

3. If the device does not maintain the reproducible state for the entire measurement period, then the 150 ms silent conditioning period may be replaced with an alternative conditioning signal (e.g. real speech signal, longer “OFF” period, etc.).

4. Apply the test signal at the levels given in Table 9 and Table 10 using the center frequencies and band limits specified in Table 11.

Table 11 – Receive SDNR Stimulus Center Frequencies and Band LimitsCenter Frequency

(Hz)Band Edges

(Hz)

400 320 480800 675 9251600 1375 18152500 2205 2855

5. Process the last 200 ms of each measured noise burst using Hanning window with 5 Hz resolution FFT from 100 Hz to 8000 Hz. After time averaging, remove the data within the notch frequency band specified in Table 12.

Table 12 – Receive SDNR Analysis Notch Frequency BandCenter Frequency

(Hz)Notch Band Edges

(Hz)

400 315 485800 670 9301600 1370 18202500 2200 2860

6. Calculate the ratio of the signal power to the total A-weighted distortion and noise power of the signal output.

7. If the measured SDNR failed and measured noise level in clause 5.2.4 was less than 10 dB from the background noise level, it may require the SDNR test to be repeated with the free field microphone repositioned at half the standard test position distance along the 50 cm axis. See Example three in clause 4.1.

5.2.7. Speakerphone Receive DelaySee 5.1.7 for more information on delay.

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The receive delay is defined here as the time from when a digitized representation of a signal arrives at that telephone’s network interface to the time its analog reproduction is received at the free field measuring microphone. The speakerphone receive delay requirements include any depacketization, hardware/firmware processing and de-jitter delays, plus any delay associated with the radio link for cordless products.

5.2.7.1. Requirement1. Corded speakerphone terminals shall be configurable so that requirements of at least Category B are

met.2. Cordless speakerphone terminals should be configurable so that the requirements of at least Category

C are met.3. The compliance category of the product shall be the same as the longest measured latency category

for send or receive. Products that do not meet category C cannot be categorized.

Table 13 – Speakerphone Receive Delay RequirementsCategory Delay Time (ms)

A ≤ 40B ≤ 85C ≤ 100

NOTE: This standard specifies delay in terms of categories for network planning purposes, similar to ITU-T Recommendation P.1010.

5.2.7.2. Measurement MethodA digital audio measuring device capable of measuring the delay between an injected digital signal and the output of the Free Field measuring microphone is connected to the network input of the telephone and to the Free Field measuring microphone. All delays inherent in the measurement system itself must be subtracted from the overall delay measurement to obtain the exact receive delay. The small delay of sound traveling over the 50 cm distance does not need to be considered.

If the telephone uses packet based transmission, then the telephone should be set to receive packets with a maximum speech frame rate of 20 ms and with one speech frame per packet.

NOTE: To minimize the delay measurement uncertainty, it is recommended to use the shortest possible packet size for the delay measurement.

A pulsed digital signal of -16 dBm0 shall be injected to the telephone’s network interface without test signal jitter. The delay between the time the test signal was injected at the telephone network interface to the time it was received at the Free Field measuring microphone is measured and used to determine the corresponding category.

5.3. Talker Sidetone ConsiderationsIf Sidetone is part of any overall application of a speakerphone system, then effective removal of that sidetone is recommended (i.e. speakerphone device connected to the handset port of a digital telephone). This may be accomplished with line echo cancellers or other such echo path suppression devices. In the rare case where some sort of sidetone is desired, then it is up to the manufacturer to adequately describe the desire performance criteria.

34

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951952953954955956957958959960961962963964965966967968969970971972973974975976


It is outside the scope of this standard to address sidetone cancellation or performance criteria.

5.4. Weighted Terminal Coupling Loss (TCLw)The weighted terminal coupling loss (TCLw) provides a measure of the echo performance under normal conversation, i.e., single far-end talker conditions. It is possible that echo control devices such as echo suppressors or echo cancellers with non-linear processing may be used on speakerphone devices to provide sufficient echo return loss to mitigate increased echo associated with longer network delays.

The use of echo control devices on a speakerphone can affect the measurement of TCLw. The result would likely be different under cases of either single far-end talker or double-talk. The TCLw measurement is intended to represent a single far-end talker. This may provide idealized and unrealistic performance measurements when non-linear processing on the transmit side is used or with a near-end signal present that is a) capable of enabling echo control’s double-talk detector with the subsequent removal of non-linear processing and b) can be filtered out from the final return signal so as not to affect the accuracy of the TCLw measurement. The latter may be the only method that can be used consistently across products in a black-box testing setup. The ‘proper’ measurement of echo is addressed in IEEE Std 269, Annex O.

5.4.1.1. RequirementsFor either the output level or the loudness rating based measurements, the normalized value of TCLw loss shall be greater than:1. 50 dB for packet-based devices2. 45 dB for TDM-based devices

NOTES:1. Calculation examples:

a. Loudness ratings based: If the measured TCLw is 49 dB with a previously measured SLR of 14 dB and RLR of 14 dB, then the normalized value of TCLw = 49 dB + (13 - 14) dB + (16-14) dB = 50 dB [TCLw measured + (13 - SLR) dB + (16 - RLR) dB].

b. Output level based: If the measured TCLw is 49 dB with a previously measured send level of 21 dBm0 and receive level of 67 dB, then the normalized value of TCLw = 49 dB + (25 - 21) dB + (67-65) dB = 55 dB [TCLw measured + (25 - level) dB + (level - 65) dB].

2. The requirement of 50 dB for packet-based devices is a function of the -20 dBm0 test signal level and the -63 dBm0(A) send noise requirement. Measuring TCLw > 50 dB can be difficult.

3. If equipped with adjustable receive level, the un-normalized TCLw will decrease in proportion with the increased gain relative to the nominal RLR in most cases. For example, if the measured TCLw is 45 dB at nominal RLR and the adjustable receive level adds 12 dB of gain, then un-normalized TCLw (maximum receive level) = 45 dB - 12 dB = 33 dB. It is therefore recommended that TCLw performance be maintained or enhanced where possible for louder volume control settings.

4. The echo impairment perceived by the person at the far-end of the connection from a telephone set is a function of the magnitude of the talker echo signal as well as the talker echo path delay. The echo signal becomes more disturbing as the talker echo path delay increases. Thus, a telephone set with adequate TCLw performance on low delay may not provide satisfactory echo loss for connections that have a long delay.

5. Temporally weighted terminal coupling loss (TCLt) is an alternate method for echo measurement, which may be more subjectively relevant, especially in devices with echo

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add "with SLR normalized to 13 dB and RLR normalized to 16 dB"


The Clause number should be 5.4.1 not 5.4.1.1


suppression or cancellation features. (See IEEE Std 1329.) The performance requirements may need to be changed when using this method.

5.4.1.2. Measurement MethodThe TCLw measurement shall be made at an input signal level of -20 dBm0.

The TCLw measurement cannot not be performed using a sinusoidal test signal for the receive path input. The test signal may be a composite source signal (CSS) as defined in ITU-T P.501 or bursted white noise as defined in IEEE Std 269. The test signal is band-limited to 100 Hz through 8000 Hz. The calibration is determined during the ON portions of the test signal, not the average of the on and off times. The measurement is performed after system stability is reached (including convergence of any echo algorithms): this is accomplished by invoking the test signal for at least 2 seconds before the actual measurement occurs.

The attenuation from digital input (receive) to digital output (send) is measured at 1/12th octave bands, using the measurement arrangement shown in Figure 5.

The weighted terminal coupling loss is calculated according to ITU-T G.122 Annex B.4 (trapezoidal rule) using the frequency range of 300 to 6700 Hz rather than 300 to 3400 Hz.

NOTES:1. ITU-T P.311 uses 100 to 8 kHz, and G.122 does not explicitly list frequencies for an

extension to wideband. Some suggest 150 to 6800 Hz would be more appropriate.2. Duplex behavior and performance concerns the ability to interrupt the other talker when they

are speaking.3. The device should have sufficient transmission loss removal in the echo control system so

that one party can interrupt the other.4. IEEE Std 1329 discusses how to measure this parameter, along with other temporal non-

linear processes, but no specified requirement for duplex performance is given in this document.

5. Subjective analysis is recommended.

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Annex A (Normative) – Speakerphone Receive Loudness Rating (RLR)RLR is the loudness loss in the receive direction from the electrical signal at the digital reference point to the acoustic signal at the free field microphone. Refer to Annex C and ITU-T Recommendation P.79.

NOTE: The nominal RLR value of +16 dB is calculated using the formula in Annex C and does not include the 14 dB correction factor used in other standards (see IEEE Std 1329 and ITU-T Recommendation P.340) speakerphone/handsfree RLR calculations.

Annex B (Normative) – Speakerphone Send Loudness Rating (SLR)The SLR is the loudness loss in the send direction from the acoustic signal at the mouth reference point to the send signal at the digital reference point. Refer to Annex C and ITU-T Recommendation P.79.

NOTES:1. This SLR target is 5 dB quieter relative to a handset due to higher talker levels, small

frequency response differences and a desire to reduce unwanted ambient noise to be transmitted to the far-end listener.

2. The nominal test position angle for a device is the direction that a single user is most likely to face. Typically, this is the STP and would be the 0 degree position.

3. The device and/or system may have Send Automatic Gain Control (AGC), that may alter the loudness rating depending on incoming levels. The AGC should not affect SLR when the STP or RTP and nominal test position angle is used, and test speech levels are at -4.7 dBPa).

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Annex C (Normative) – Calculation of Loudness RatingsThis Annex details the loudness rating calculations and weighting factors relevant to the TIA-920.xxx-A series of documents. Loudness ratings are a measure of loudness loss and are used in network planning to insure that the loudness of a connection from the MRP of the talker to the ERP of the far-end listener is at a satisfactory level. The loudness of the complete path is designated as the wideband Overall Loudness Rating (OLR). The wideband SLR is the loudness loss from the MRP to the electrical output. The wideband RLR is the loudness loss from the electrical input to the ERP.

Loudness ratings are used rather than simple level measurements because of better subjective correlation. Loudness ratings more closely account for the ear’s different sensitivity at different frequencies and its non-linear response to varying sound levels. The following calculations are based on the ITU-T Recommendation P.79. ITU-T Recommendation P.79 provides information on the derivation of the loudness rating algorithm.

Loudness ratings determined in accordance with P.79 are analogous to loss, with more positive numbers representing greater loss or less loudness from the telephone.

ITU-T Recommendation P.79 Annex A, Table A.2 has the weighting factors used for wideband SLR and RLR calculation. For convenience, the P.79 weighting information is included in this document in TableC.1.

C.1 Wideband Send Loudness Rating (Handset, Speakerphone and Headset)Equation C-1:

Band 20

SLR = - 57.1 log10 10(0.1 * 0.175 * (SMJ – Wsi ))

i = Band 1Where:

i Frequency bands from Table C.1, bands 1-20.SMJ Send frequency response data (Sensitivity, Mouth-to-Junction) in dBV/Pa measured per

this standard.Wsi Send weighting factor from Table C.1.

C.2 Wideband Receive Loudness Rating (Speakerphone only)Equation C-2:

Band 20

RLR = - 57.1 log10 10(0.1 * 0.175 * (SJE – Wri ))i = Band 1

Where: i Frequency bands from Table C.1, bands 1-20.SJE Receive frequency response data (Sensitivity, Junction to 50cm Reference Point as

defined for STP) in dBPa/V measured per this standard. Wri Receive weighting factor from Table C.1.

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Table C.1 – Weighting Factors for Calculating Wideband Loudness Ratings and STMR

(Ref. ITU-T P.79 Annex A, Table A.2 and Table B.2)Band No. Mid-frequency

(Hz)SendWsi

ReceiveWri

Sidetone WMSi

1 100 154.5 152.8 110.42 125 115.4 116.2 107.73 160 89.0 91.3 104.64 200 77.2 85.3 98.45 250 62.9 75.0 94.06 315 62.3 79.3 89.87 400 45.0 64.0 84.88 500 53.4 73.8 75.59 630 48.8 69.4 66.0

10 800 47.9 68.3 57.111 1000 50.4 69.0 49.112 1250 59.4 75.4 50.613 1600 57.0 70.7 51.014 2000 72.5 81.7 51.915 2500 72.9 76.8 51.316 3150 89.5 93.6 50.617 4000 117.3 114.1 51.018 5000 157.3 144.6 49.719 6300 172.2 165.8 50.020 8000 181.7 166.7 52.8

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Annex D (Informative) – Measurement and Level Conversions

The following describes how to convert between various units of measurement used in telephone testing.

0 dBm (0 VU) is accepted as 1 mW, typically using a circuit impedance of 600 or 900 .

0 dBm = 10 log 1(mW)

dBV = 10 log V2 or, V = 10 dBV/20= 20 log V

P = V2/R, where for dBm reference, R = 600

dBm = 10 log (V2/R * 1000)= 10 log (V2/600 * 1000)= 10 log (V2/0.600)

Therefore, for 0 dBm, V = 774.6 mV or 0 dBm = -2.218 dBV @ 600 (use -2.2 dB)

P = V2/R, where for dBm reference, R = 900

dBm = 10 log (V2/R * 1000)= 10 log (V2/900 * 1000)= 10 log (V2/0.900)

Therefore, for 0 dBm, V = 948.7 mV or 0 dBm = -0.458 dBV @ 900 (use -0.5 dB)

This means that if we substitute 600 for 900 or vice versa, and the voltage remains constant, then we have:

Correction (dB) = -10 log 0.600/0.900 = 10 log 0.900/0.600 = 1.761 dB

To simplify,Correction (dB) = 10 log( |Z1| / |Z2| ), that is, the log of the ratio of the magnitude of the impedances, when converting from impedance Z1 to Z2.

If converting from "Z1 = 600 " to "Z2 = 900 ", the correction factor is -1.76 dB (use -1.8 dB), therefore subtract 1.8 dB from the measurement.

At this point, depending on the impedance, conversion factors can be applied dB for dB to the measured or calculated result. For example, to convert a 600 , -20 dBm signal to dBV, subtract 2.2 to get -22.2 dBV. Another example is if -20 dBm is measured across 600 , then when measuring across 900 , add a correction of -1.8 dB to get -21.8 dBm (since less power is dissipated by the higher resistance).

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Annex E (Informative) – R40 Preferred FrequenciesThe ISO 3, R40 basic series preferred frequencies are listed in Table E.1. The frequencies highlighted in Italics are the R10 series of preferred frequencies. The R40 series of preferred frequencies is based on 1/12th octave frequencies, but the numbers are rounded in a convenient pattern. The R10 series is based on the 1/3rd octave frequencies.

Table E.1 – ISO R40 and R10 Preferred Frequencies, 1/12th and 1/3rd Octave Frequencies

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42

R40 PreferredFrequencies, Hz.

1/12 Oct. BandCenter Freq, Hz.

1/3 Oct. BandCenter Freq, Hz.

R10 PreferredFrequencies, Hz.

9091.73

100.00 100

9597.16

100102.92

106109.02

112115.48

125.89 125118 122.32125

129.57132137.25

140145.38

158.49 160

150153.99160163.12

170172.78

180183.02

199.53 200

190193.87

200205.35

212217.52224230.41

251.19 250236

244.06250

258.52265273.84

280290.07

316.23 315300

307.26315325.46

335 344.75355

365.17

398.11 400375

386.81400 409.73425

434.01450459.73

501.19 500475 486.97500

515.82530546.39560 578.76

630.96 630600 613.06630 649.38670 687.86710 728.62

794.33 800750 771.79800 817.52850 865.96900 917.28

1000.00 1000950 971.631000 1029.201060 1090.18

11201120 1154.78

1258.93 1250

1180 1223.211250

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Annex F (Informative) – Terms Related to Digital Transmission Levels

dBm: (sometimes also called dBmW) is an abbreviation for the power ratio in decibels (dB) of the measured power referenced to one milliwatt (mW). The logarithmic expression for power, referenced to 1 milliwatt is dBm = 10 log (power / 1 mW). For electrical power level measurements, 0 dBm can also be referred to as the power of 0.775 volts RMS Sinusoidal signal applied across a defined resistive load (often 600 Ohms in telecommunications), which results in a reference active power level of 1 mW. Note: specifying dBm without specifying the impedance may be misleading. For digital networks, since there is no impedance used, it should also be defined by the corresponding Sinusoidal peak code value (dBFS, or dBov) for the given codec.

dBov/dBFS: dBov stands for dB overload. It is the rms signal level of a digital signal relative to the overload or maximum peak level of the digital format. This is also commonly referred to as dBFS (Full Scale). For example, a rectangular function with only the positive or negative maximum number is 0 dBov; a single frequency tone with peak at maximum level is -3.01 dBov. Note: 0 dBm0 equals (-3.17-3.01) = -6.18 dBFS.

TLP-Max: Maximum Transmission Level Point is defined as the maximum dBm power measured across a defined impedance (e.g. 600 Ohm resistive) by applying a full scale (0 dBFS or dBov) Sinusoidal signal (just below its clipping point). Consequently, Max-TLP for A-law and μ-law codecs in a system, are defined as +3.14 dBm and +3.17 dBm, respectively.

0-TLP: Zero Transmission Level Point is an arbitrarily established point in a system relative to which transmission levels at all other points in the system are measured. 0-TLP for A-law and μ-law codecs in a system is defined as -3.14 dB and -3.17 dB, below TLP-Max, respectively. In digital transmissions networks, the 0-TLP is often referred to as 0 dBm0, which is also equivalent to 0 dBm).

dBm0: Is defined as the power level in dBm relative to a reference point called the Zero Transmission Level Point, or 0-TLP. This level for A-law and μ-law codecs is defined as -3.14 dB and -3.17 dB below the full scale (or TLP-Max.), respectively. Consequently, for an A-law and μ-law codecs, the 0 dBm0 (which is also referred to as 0 digital reference point or digital milliwatt) and 0-TLP are all the same, and may be used interchangeably. Also note that 0-TLP, sometimes is referred to as 0 dBr (or dB relative to 0 dBm) in some earlier telecom literature.

For the purposes of this standard, 0 dBm0 (or 0-TLP) is the RMS level of a Sinusoidal wave with -3.17 dB below L16-256 TLP-Max level, and -9 dB below G.722 codecs TLP Max level. Consequently, the 0 dBm0 digital reference point may be defined for 0 dBm Sinusoidal test signal considering the following TLP-Max designations:

Codec Type Defined TLP-Max in dBmat 600 Ohm

0 dBm0 reference pointis defined as 0 dBm at

A-law +3.14 (TLP-Max) – 3.14

μ-law +3.17 (TLP-Max) – 3.17

L16-256 +3.17 (TLP-Max) – 3.17

G.722 +9.00 (TLP-Max) – 9.0

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NOTE: It is the responsibility of the transcoder to correctly implement the digital full scale to dBm0 level translation for each codec when digital signals are translated from one codec format to another, and to ensure that signal levels at one port of the transcoder are correctly translated to the appropriate signal levels at the other port of the transcoder.

This simply means that a continuity test tone (1004 Hz) entering one end of the network at 0 dBm level (terminated with 600 Ohm impedance), should produce a 0 dBm output signal across 600 Ohm load on the other end of the network independent of any codec types or transcoding used in between the two points.

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Annex G (Informative) - Recommendations for subjective quality evaluationAll devices should be evaluated subjectively during a regular call. The goal is to identify any obvious problems that may have been missed throughout the objective tests. The evaluation should be performed under normal usage condition and some stress conditions. If the device requires a softphone client to make a call, the evaluation below should be performed with the device connected to a softphone client that it is intended to be used with. In the testing scenarios described below, the term device under test (DUT) is used to refer to both standalone device endpoints and to endpoints formed by the peripheral device connected to a softphone client. All tests below require two people. The term near-end user describes the person using the DUT and the term far-end user describes the person in a remote room who is using a reference device to communicate with the near-end user.

Normal usage conditions: Moderately sized room, typically at least 12 square meters with reverberation time of 0.5 to 1.00

seconds in the speech band. Room ambient noise level: 40-45dBA. Hoth noise generated from a loudspeaker facing a corner

of the room can be used to produce this level of noise. Human speech at normal level, typically, 60 to 70dBSPL measured at 0.5 meters from the mouth. Receive volume control is set to a comfortable listening level.

Examples of stress conditions: Moderately sized room, typically at least 12 square meters with reverberation time of 0.5 to 1.00

seconds in the speech band. Room ambient noise level: 50-55dBA. Hoth noise generated from a loudspeaker facing a corner

of the room can be used to produce this level of noise. Human speech at quiet level, typically 55-60 dBSPL, and loud level, typically 80-85 dBSPL,

measured at 0.5 meters from the mouth. Receive volume control is set to maximum level.

Far-end setup conditions: The far-end talker is located in a quiet office environment with a high quality reference corded

handset or corded monaural headset. The conformance of this handset or headset to TIA-920.110A or TIA-920.130A respectively should be verified. Only monaural headsets should be used for this test. The loudness setting on the handset/headset should match the nominal receive loudness rating (RLR) as specified in TIA-920.110A/TIA-920.130A.

Near-end setup conditions: Review the DUT user manual for the usage scenarios that are recommended by the manufacturer.

The near-end talker uses the product in a real meeting room/office as instructed in the user manual. If there isn’t any manufacturer’s specification then similar positioning should be used as described in IEEE Std 1329.

G.1 Send loudness

PurposeThis evaluation is performed to verify the DUT send signal level is sufficient when received by the far-end. This evaluation should also help reveal any problems with built-in gain control.

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RecommendationThe DUT send signal should be loud enough for the far-end user to hear comfortably. This recommendation should be met at the standard or recommended test position with a normal and quiet talker level.

Evaluation Procedure1. Set up a call between the two participants.2. Start the evaluation by the near-end talker speaking into the DUT at the recommended test

position.3. The far-end talker evaluates if the near-end talker is easily understood.

If the near-end talker sounds too soft, then the send loudness of the DUT is not sufficient. If the voice level received by the far-end talker changes unexpectedly during the sentences, it

may indicate a problem in the DUT’s send gain control mechanism.4. Repeat step 3 with the near-end talker moving away from the DUT to the border of the DUT’s

maximum supported microphone pick-up range.

G.2 Send noise

PurposeThis evaluation is performed to verify that no excessive noise is transmitted from the DUT to the far-end listener. It will also verify that there are no disruptive noise level changes often referred to as “noise pumping”

RecommendationThis evaluation should be performed in both normal and high noise level conditions.

Under normal noise conditions, the near-end user’s voice should be transmitted from the DUT without objectionable noise. If the near-end user is silent then there should be no objectionable noise sent by the DUT.

Under high noise conditions the far-end listener may hear elevated level of background noise from the near-end user. However, the noise level should not vary with the speech level.

Evaluation procedure1. Set up a call between the two participants.2. Start the evaluation by the near-end talker speaking into the DUT at the recommended test

position with silence periods of 10 seconds or more in between sentences.3. The far-end talker should listen for any excessive noise. Also the far-end talker should listen for

any noticeable noise level change. If the far-end talker hears objectionable noise or noise pumping effects, it may indicate a problem with the DUT’s noise suppression features.

4. Repeat steps 2 and 3 with high ambient noise condition.

G.3 Send distortion

PurposeThis evaluation is performed to verify that when the near-end user is speaking in a normal or loud level, the far-end listener does not perceive distortion due to microphone clipping of the DUT.

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RecommendationWhen the near-end user is talking into the DUT in a normal level, the far-end user should not hear any objectionable distortion. Similarly in a stress condition with a loud near-end level, the far-end user should also not hear any objectionable distortion. Distortion may be caused by clipping of the microphone signal.

Evaluation procedure1. Set up a call between the two participants.2. While the far-end talker is talking at a normal level, the near-end talker sets the receive volume to

their preferred listening level.3. When the near-end talker speaks with a normal level, the far-end listener should not hear any

objectionable distortion.4. When the near-end talker speaks with a loud level, the far-end listener should not hear any

objectionable distortion.

G.4 Acoustic echo cancellation performance during single-talk

PurposeThis evaluation is performed to verify there are no objectionable echo leaks when the far-end is talking. Additionally when testing with alternating speech from near-end and far-end user there should be no objectionable distortion when the near-end talker is talking.

RecommendationWhile the far-end is talking to the near-end, the far-end should not hear any far-end speech echoing back. The echoes can be either full echo of the speech or residual echo leaks that may sound like noise correlated with the far-end speech.

During alternating speech the far-end listener should also not hear any objectionable speech distortion in the near-end talker’s speech. This should be verified under normal conditions where the receive loudness setting of the DUT is set such that the nominal receive loudness is achieved as well as the near-end talker at nominal level. This should then also be verified under stress conditions with the receive volume set to maximum as well as the near-end talker quiet level.

Evaluation procedure1. Set up a call between the two participants.2. The near-end talker sets the receive volume to achieve a comfortable listening level.3. Start the evaluation by having the near-end and the far-end take turns speaking.4. The far-end listens to whether his own voice or a distorted version of his own voice is echoing

back. Additionally, the far-end should pay attention to the voice quality of the near-end’s voice.5. Repeat steps 3 and 4 for the maximum receive volume supported by the DUT.6. Repeat steps 2 through 4 with quiet level speech at the near-end. Adjust the receive volume

setting of the DUT to achieve a comfortable listening level.7. Repeat step 6 for the maximum receive volume supported by the DUT.

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G.5 Acoustic echo cancellation performance during double-talk

PurposeThis evaluation is performed to verify that during a normal cadenced conversation the near-end and the far-end can easily interject while the other end is talking, and also verify there are no echo leaks in double-talk situations.

RecommendationThe far-end should not hear any objectionable echo leaks during double-talk operation. Slight signal distortion of the near-end speech signal is acceptable during double talk. The near-end should be able to interrupt the far-end and the far-end should be able to interrupt the near-end. This should be verified under normal conditions where the receive loudness setting of the DUT is set such that the nominal receive loudness is achieved as well as where the near-end is at the nominal level. This should then also be verified under stress conditions with the receive loudness setting set to maximum as well as the near-end is at the quiet level.

Evaluation procedure1. Set up a call between the two participants.2. The near-end talker sets the receive volume to achieve a comfortable listening level.3. Start the evaluation by taking turns talking for 10 seconds. Gradually try more and more double-

talk with the near-end and the far-end interrupting each other. This should include different duration of pauses prior to interruptions. The interrupting user should try interruptions after short pauses (1-2 seconds) and after longer pauses (approx. 10 seconds). Both the near-end and far-end should perform interruption tests. The evaluation may also include the extreme case of consistent double-talk where one participant may count numbers and the other participant may recite the alphabet or read continuously. For the test with consistent double-talk, it is expected that only full-duplex type 1 devices as defined in IEEE Std 1329 will provide sufficient interruptibility whereas type 2 and type 3 devices may provide only reduced interruptability.

4. The far-end listens to verify whether his own voice or a distorted version of his own voice is echoing back.

5. The far-end talker evaluates if it was possible to easily interrupt the near-end talker. If it was not possible to easily interrupt then the DUT is identified as not providing sufficient interruptibility during send operation.

6. The near-end talker evaluates if it was possible to easily interrupt the far-end talker. If it was not possible to easily interrupt then the DUT is identified as not providing sufficient interruptibility during receive operation.

7. Repeat steps 3 through 6 for the maximum receive volume supported by the DUT.8. Repeat steps 3 through 6 with quiet level speech at the near-end. Adjust the receive volume

setting to achieve a comfortable listening level.9. Repeat step 8 for the maximum receive volume supported by the DUT.

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Annex H (Informative) – BNR StudyThis annex is a version of an internal memo on the results for preferred listening levels for handsfree terminals (speakerphones), completed in the BNR (now Nortel) Subjective Assessment Lab in August 1986. This study examined user preference for speakerphone receive output levels in quiet and with several levels of room noise. Recommendations for speakerphone nominal level were given based on user response to specific levels across a range of room noise values.

Nortel provided this data in support of the development of TIA-470.120-C. These results were originally contributed to IEEE in 1995 in support of similar work, and they are consistent with TIA and ITU recommendations for the electro-acoustic levels of speakerphones.

H.1 Objectives

1. To determine preferred listening levels for loudspeaker listening.

2. To examine the relationship between preferred listening .level and room noise.

H.2 ApparatusA loudspeaker was placed on the table 31.5 cm behind the standard handsfree position, and tilted 10 degrees upwards. Acoustically transparent foam hung, from ceiling to table height, in front of the loudspeaker. A disconnected Northern Telecom Companion 4 terminal was placed in the standard experimental position (40 cm in front of subject). The speech samples were bandpass filtered at 500-3500 Hz to simulate the telephony speech band and equalized (per sentence) for signal level. Speech levels were measured using a B&K sound pressure meter that fed into a BT Speech Voltmeter SV6. Measurements were made using a 60 second sample of recorded speech (same samples and talkers used in the study). Levels were measured from the standard handsfree measuring position, 40 cm in front, and 30 cm above the handsfree terminal. The speaker amplifier was calibrated to produce an unattenuated speech level of 89.5 dB SPL. A programmable attenuator was used to control the speech level heard on each trial.

H.3 Procedure Subjects sat facing the loudspeaker, with the chair placed at a natural distance from the table. A speech sample consisting of one sentence was played through the loudspeaker. Although speech samples were presented through the loudspeaker, subjects reported that speech appeared to come from the handsfree terminal.

Subjects were asked to rate whether speech samples were "Too Soft' , "OK", or "Too Loud", and indicate their response using the button box. Twelve trials were randomly presented at each of the ten speech levels shown in the table below. Three blocks of trials were given to each subject, one at each of the three room noise levels: 35, 45, and 55 dBA. Blocks occurred in random order. The 360 trials took approximately 20 minutes to complete. Results below are based on responses from 20 subjects.

H.4 Preliminary Results As room noise increases, preferred listening level increases. At 35 dBA the preferred listening level is 70 dB SPL, at 45 dBA it is 73 dB SPL, and at 55 dBA preferred listening level is 76 dB SPL. The rate of increase of preferred listening level with increasing room noise is approximately .3 dB/dB, which is in line with Gardner's (1969) and Ericsson's (1977) results.

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There appears to be a threshold between 79 and 82 dBSPL where speech is considered too loud, regardless of the room noise level.

Compression of the acceptable range of speech levels as room noise increases, as reported by Gardner and others, is not readily apparent in the present results. In all room noise conditions, there are at least three speech levels where %OK is above 80%. Substantial compression may not occur until higher room noise levels, or maybe the compression information is lost when data is averaged.

H.5 RecommendationsA good compromise for nominal level appears to be 73 dB SPL, since it gives over 90% acceptance (94,97,95) at all three noise levels. An acceptable range for volume would be 61 to 82 dB SPL; 64% found 61 dB to be too low even in the quiet room, and 55% found 82 dB to be too loud in the noisiest room. The step size should be smaller than 3 dB, since a single 3 dB step can change opinion by over 40%.

It should be remembered that the present results were obtained in the large sound room, which is not a very reverberant room. Since sound is absorbed readily in this room, subjects may prefer higher listening levels in this room than in a more reverberant room. This should be taken into account when generalizing these results to other environments.

H.6 Table of ResultsThe following are the acceptance of speech levels for three levels of room noise.

Table H.1 – Room Noise 35 dBA User Acceptance LevelsRoom Noise = 35dBA N=20Speech Level (dBSPL) % Too Low % OK % Too High

85 0 2 98

82 0 12 87

79 0 37 63

76 0 74 25

73 0 94 5

70 0 99 0

67 7 93 0

64 34 64 1

61 64 35 0

58 87 12 0

50

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85 0 5 94

82 1 24 74

79 0 61 38

76 1 88 10

73 1 97 1

70 4 95 0

67 25 74 0

64 66 33 0

61 86 13 0

58 97 2 0


85 0 13 86

82 0 44 55

79 0 75 25

76 3 96 1

73 4 95 0

70 20 80 0

67 60 39 0

64 88 12 0

61 97 3 0

58 100 0 0

5152

1456

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Annex I (Informative) – BibliographyThe following standards are identified for informational purposes only and do not constitute provisions of this Standard. At the time of publication, the editions indicated were valid. All standards are subject to revision, and users of this Standard may want to consider the most recent editions of the standards indicated below when looking for further information. ANSI and TIA maintain registers of currently valid national standards published by them.

1. ANSI/TIA-470.120-C-2011, Telecommunications Telephone Terminal Equipment Transmission Requirements for Analog Speakerphones.

2. ANSI/TIA-571-B-2007, Telecommunications Telephone Terminal Equipment Electrical, Thermal, Mechanical Environmental Performance Requirements.

3. ANSI/TIA-810-B-2006, Telecommunications Telephone Terminal Equipment Transmission Requirements for Narrowband Digital Telephones.

4. IETF RFC3551 (2003), (L16-256 codec) RTP Profile for Audio and Video Conferences with Minimal Control.

5. ITU-T Recommendation P.51 (1996-08), Artificial mouth.6. ITU-T Recommendation P.57 (2009-04), Artificial ears.7. ITU-T Recommendation P.58 (1996-08) and P.58 Note (2008), Head and torso simulator for

telephonometry.8. ITU-T Recommendation P.311 (2011-03), Transmission characteristics for wideband (150-7000

Hz) digital handset telephones.9. ITU-T Recommendation P.340 (2000-05), Transmission characteristics for hands-free telephones.10. ITU-T Recommendation P.360 (2006-07), Efficiency of devices for preventing the occurrence of

excessive acoustic pressure by telephone receivers.11. ITU-T Recommendation P.1010 (2004-07), Fundamental voice transmission objectives for VoIP

terminals and gateways.12. ITU-T Recommendation G.712 (2001-11), Transmission performance characteristics of pulse code

modulation channels.

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