ETSI TS 102 925 V1.2.1 (2018-10)

Speech and multimedia Transmission Quality (STQ); Transmission requirements for Super-Wideband / Fullband handsfree and conferencing terminals from a QoS perspective as perceived by the user

RTS/STQ-208-2

General Information

• Status: Published
• Publication Date: 17-Oct-2018
• Technical Committee: STQ - Speech and multimedia Transmission Quality
• Current Stage: 12 - Completion
• Due Date: 23-Oct-2018
• Completion Date: 18-Oct-2018

Standards Content (Sample)

TECHNICAL SPECIFICATION
Speech and multimedia Transmission Quality (STQ);
Transmission requirements for Super-Wideband / Fullband
handsfree and conferencing terminals from
a QoS perspective as perceived by the user

2 ETSI TS 102 925 V1.2.1 (2018-10)

Reference
RTS/STQ-208-2
Keywords
QoS, terminal
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3 ETSI TS 102 925 V1.2.1 (2018-10)
Contents
Intellectual Property Rights . 5
Foreword . 5
Modal verbs terminology . 5
Introduction . 5
1 Scope . 6
2 References . 6
2.1 Normative references . 6
2.2 Informative references . 8
3 Definitions of terms and abbreviations. 9
3.1 Terms . 9
3.2 Abbreviations . 9
4 Applications and Codec considerations. 10
4.1 Applications . 10
4.2 Codec considerations . 10
4.2.0 Introduction. 10
4.2.1 Super-wideband (SWB) . 11
4.2.2 Fullband (FB). 12
5 Test equipment and associated considerations . 13
5.0 Introduction . 13
5.1 Test Set-up. 13
5.1.0 Introduction. 13
5.1.1 Setup for terminals . 14
5.1.1.0 Introduction . 14
5.1.1.1 Desktop operated handsfree terminal . 14
5.1.1.2 Handheld handsfree terminal . 15
5.1.1.3 Softphone (computer-based terminals) . 16
5.1.1.4 Group audio terminal (GAT). 19
5.1.1.5 Teleconference systems . 21
5.1.1.6 Void. 21
5.1.2 Test signals . 21
5.1.3 Test signal levels . 21
5.1.3.0 General . 21
5.1.3.1 Send. 21
5.1.3.2 Receive . 22
5.1.4 Setup of background noise simulation . 22
5.1.5 Acoustic environment . 23
5.1.5.1 Measurement environment . 23
5.1.5.2 Acoustic environment for the rooms where systems are implemented . 23
5.1.6 Influence of terminal delay issue on measurements . 23
5.2 Environmental conditions for tests . 24
5.3 Accuracy of measurements and test signal generation . 24
5.4 Specific test considerations . 25
5.4.0 Introduction. 25
5.4.1 Loudness Rating and Loudness . 25
5.4.1.1 Loudness Rating . 25
5.4.1.2 Loudness . 25
5.4.2 Binaural listening . 26
5.4.3 Subjective considerations . 26
5.4.4 Setup of variable echo path . 26
5.5 Network impairment simulation . 27
5.6 Verification of the environmental conditions . 28
6 Requirements and associated measurement methodologies . 28
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4 ETSI TS 102 925 V1.2.1 (2018-10)
6.0 Considerations . 28
6.1 Send . 28
6.1.0 Introduction. 28
6.1.1 Frequency response. 28
6.1.2 Send Loudness rating (SLR) . 30
6.1.3 Void . 31
6.1.4 Send Noise . 31
6.1.5 Send Distortion . 31
6.1.5.1 Signal to harmonic distortion versus frequency . 31
6.1.5.2 Signal to harmonic distortion for higher input level . 33
6.1.6 Microphone mute . 33
6.2 Receive . 33
6.2.0 Introduction. 33
6.2.1 Equalization . 33
6.2.2 Frequency response. 34
6.2.2.0 General . 34
6.2.2.1 Handheld terminal . 34
6.2.2.2 Desktop terminal . 36
6.2.2.3 Terminals intended to be used simultaneously by several users . 37
6.2.3 Receive Loudness Rating (RLR) and Loudness . 38
6.2.3.1 Receive Loudness Rating . 38
6.2.3.2 Loudness . 39
6.2.4 Receive Noise . 39
6.2.5 Receive Distortion . 40
6.3 Other parameters . 41
6.3.0 Introduction. 41
6.3.1 Round-trip Delay . 41
6.3.1.1 Round-trip Delay for VoIP terminals . 41
6.3.1.2 Quality of jitter buffer adjustment . 43
6.3.2 Terminal Echo Loss (TCL) . 44
6.3.3 Objective listening quality . 44
6.3.4 Double talk performance . 45
6.3.4.1 General . 45
6.3.4.2 Attenuation range in send direction during double talk A . 45
H,S,dt
6.3.4.3 Attenuation range in receive direction during double talk A . 47
H,R,dt
6.3.4.4 Detection of echo components during double talk . 47
6.3.4.5 Minimum activation level and sensitivity of double talk detection . 49
6.3.5 Speech and audio quality in presence of noise. 50
6.3.5.1 Performance in send in the presence of background noise . 50
6.3.5.2 Speech quality in the presence of background noise . 50
6.3.5.3 Quality of background noise transmission (with far end speech). 51
6.3.6 Potential other quality features . 52
6.3.6.1 Sound localization and binaural performance . 52
6.3.6.2 Dereverberation performance . 52
6.3.6.3 Switching characteristics between transducers . 52
6.3.7 Quality of echo cancellation . 52
6.3.7.1 Temporal echo effects . 52
6.3.7.2 Spectral echo attenuation . 53
6.3.7.3 Occurrence of artefacts . 53
6.3.7.4 Variable echo path. 53
6.3.8 Switching characteristics . 54
6.3.8.1 Note . 54
6.3.8.2 Activation in send direction . 54
6.3.8.3 Silence suppression and comfort noise generation . 54
Annex A (normative): Room acoustics and electro acoustic equipment positioning . 55
Annex B (informative): Bibliography . 56
History . 57

ETSI
5 ETSI TS 102 925 V1.2.1 (2018-10)
Intellectual Property Rights
Essential patents
IPRs essential or potentially essential to normative deliverables may have been declared to ETSI. The information
pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found
in ETSI SR 000 314: "Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in
respect of ETSI standards", which is available from the ETSI Secretariat. Latest updates are available on the ETSI Web
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Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee
can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the updates on the ETSI Web
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Trademarks
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ETSI claims no ownership of these except for any which are indicated as being the property of ETSI, and conveys no
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Foreword
This Technical Specification (TS) has been produced by ETSI Technical Committee Speech and multimedia
Transmission Quality (STQ).
Modal verbs terminology
In the present document "shall", "shall not", "should", "should not", "may", "need not", "will", "will not", "can" and
"cannot" are to be interpreted as described in clause 3.2 of the ETSI Drafting Rules (Verbal forms for the expression of
provisions).
"must" and "must not" are NOT allowed in ETSI deliverables except when used in direct citation.
Introduction
Speech terminals are currently implementing narrowband and wideband bandwidth. Nowadays, terminal equipment
may offer wider bandwidth, due to features already available in these terminals. Such equipment may implement
conversational features that may be to the benefit of the electro acoustic equipment's already available in the terminal
and may provide wider quality for the end users. High quality conferencing systems may also implement wider
bandwidth in order to reach quality and behaviour close to normal face to face conditions.
The present document is intended to provide initial requirements and test methods for such equipment. The present
document also provides materials for a further update of ETSI SR 002 959 [i.2]: Electronic Working Tools; Roadmap
including recommendations for the deployment and usage of electronic working tools in the ETSI standardization
process.
The present document complements the ETSI TS 102 924 [17] Handset and Headset mode specifications.
ETSI
6 ETSI TS 102 925 V1.2.1 (2018-10)
1 Scope
The present document provides speech & audio transmission performance requirements and measurement methods for
handsfree functions of super-wideband/fullband terminals, including conferencing terminals. The present document
provides requirements in order to optimize the end to end quality perceived by users.
Users become more sensitive to voice and music quality (for music used in conversational services) when using
ICT/terminal equipment and so are more demanding for further enhancement especially further extension of the audio
coded bandwidth.
For instance, this is the case for high quality conferencing services with music on hold, better background environment
rendering and longer duration than normal point to point calls.
Standardized super-wideband and fullband codecs are now available, some being also compatible with wideband
codecs.
The present document will consider only conversational services (that may be mixed with other services) and does not
cover the streaming-only services.
Such applications include:
• Speech and audio communication including conferencing using high quality handsfree systems.
• Bandwidth extension which may allow usage for some mixed content applications.
• Super-wideband enhancement coupled with stereo/dichotic/multichannel.
In the send path the signal may combine speech, music and environmental signals. The signal may be:
• acoustically captured by a microphone; or
• directly inserted through a digital or analog connection.
In the receive path, the signal may combine:
• communication signals such as described for send path, including environmental signals; and
• signals coming from distributed applications (e.g. advertisement, music on hold, etc.).
2 References
2.1 Normative references
References are either specific (identified by date of publication and/or edition number or version number) or
non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the
referenced document (including any amendments) applies.
Referenced documents which are not found to be publicly available in the expected location might be found at
https://docbox.etsi.org/Reference/.
NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee
their long term validity.
The following referenced documents are necessary for the application of the present document.
[1] Recommendation ITU-T P.501: "Test signals for use in telephonometry".
[2] Recommendation ITU-T P.10/G.100: "Vocabulary for performance and quality of service".
[3] Recommendation ITU-T P.58: "Head and torso simulator for telephonometry".
ETSI
7 ETSI TS 102 925 V1.2.1 (2018-10)
[4] Recommendation ITU-T P.581: "Use of head and torso simulator (HATS) for hands-free and
handset terminal testing".
[5] Recommendation ITU-T P.79: "Calculation of loudness ratings for telephone sets".
[6] Recommendation ITU-T P.340: "Transmission characteristics and speech quality parameters of
hands-free terminals".
[7] Recommendation ITU-T G.722.1 (Annex C): "Low-complexity coding at 24 and 32 kbit/s for
hands-free operation in systems with low frame loss".
[8] Recommendation ITU-T G.729.1 (Annex E): "G.729-based embedded variable bit-rate coder: An
8-32 kbit/s scalable wideband coder bitstream interoperable with G.729".
[9] Recommendation ITU-T G.718 (Annex B): "Frame error robust narrow-band and wideband
embedded variable bit-rate coding of speech and audio from 8-32 kbit/s".
[10] Recommendation ITU-T G.719: "Low-complexity, full-band audio coding for high-quality,
conversational applications".
[11] ETSI ES 202 740: "Speech and multimedia Transmission Quality (STQ); Transmission
requirements for wideband VoIP loudspeaking and handsfree terminals from a QoS perspective as
perceived by the user".
[12] ETSI TS 103 740: "Speech and multimedia Transmission Quality (STQ);Transmission
requirements for wideband wireless terminals (handsfree) from a QoS perspective as perceived by
the user".
[13] ETSI ETS 300 807: "Integrated Services Digital Network (ISDN); Audio characteristics of
terminals designed to support conference services in the ISDN".
[14] Recommendation ITU-T P.863: "Perceptual objective listening quality assessment".
[15] Recommendation ITU-T G.711.1: "Wideband embedded extension for G.711 pulse code
modulation".
[16] Recommendation ITU-T P.1301: "Subjective quality evaluation of audio and audiovisual
multiparty telemeetings".
[17] ETSI TS 102 924: "Speech and multimedia Transmission Quality (STQ); Transmission
requirements for Super-Wideband / Fullband handset and headset terminals from a QoS
perspective as perceived by the user".
[18] Void.
[19] Void.
[20] Recommendation ITU-T G.722: "7 kHz audio-coding within 64 kbit/s".
[21] Recommendation ITU-T P.56: "Objective measurement of active speech level".
[22] IEC 61260-1: "Electroacoustics - Octave-band and fractional-octave-band filters - Part 1:
Specifications".
[23] ISO 3745: "Acoustics -- Determination of sound power levels and sound energy levels of noise
sources using sound pressure -- Precision methods for anechoic rooms and hemi-anechoic rooms".
[24] Void.
[25] ETSI TS 126 441: "Universal Mobile Telecommunications System (UMTS); LTE; Codec for
Enhanced Voice Services (EVS); General overview (3GPP TS 26.441)".
[26] ETSI TS 103 281: "Speech and multimedia Transmission Quality (STQ); Speech quality in the
presence of background noise: Objective test methods for super-wideband and fullband terminals".
[27] Recommendation ITU-T P.863.1: "Application Guide for Recommendation ITU-T P.863".
ETSI
8 ETSI TS 102 925 V1.2.1 (2018-10)
[28] ETSI TS 103 224: "Speech and multimedia Transmission Quality (STQ); A sound field
reproduction method for terminal testing including a background noise database".
[29] Recommendation ITU-T G.122: "Influence of national systems on stability and talker echo in
international connections".
[30] IETF RFC 6716: "Definition of the Opus Audio Codec".
[31] Recommendation ITU-T P.502: "Objective test methods for speech communication systems using
complex test signals".
[32] Recommendation ITU-T P.1010: "Objective test methods for speech communication systems
using complex test signals".
2.2 Informative references
References are either specific (identified by date of publication and/or edition number or version number) or
non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the
referenced document (including any amendments) applies.
NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee
their long term validity.
The following referenced documents are not necessary for the application of the present document but they assist the
user with regard to a particular subject area.
[i.1] ITU-T Supplement P16: "Guidelines for placement of microphones and loudspeakers in telephone
conference rooms and Group Audio Terminals (GATs)".
[i.2] ETSI SR 002 959: "Electronic Working Tools; Roadmap including recommendations for the
deployment and usage of electronic working tools in the ETSI standardization process".
[i.3] ISO 532: "Acoustics - Method for calculating loudness level".
[i.4] ETSI EG 202 425: "Speech Processing, Transmission and Quality Aspects (STQ); Definition and
implementation of VoIP reference point".
TM
[i.5] NIST Net .
NOTE: Available at https://www-x.antd.nist.gov/itg/nistnet/.
TM
[i.6] Netem .
NOTE: Available at http://www.linuxfoundation.org/en/Net:Netem.
[i.7] STQ(15)48_039: "Objective Codec Evaluation of EVS. HEAD acoustics GmbH".
[i.8] ETSI TR 126 952: "Universal Mobile Telecommunications System (UMTS); LTE; Codec for
Enhanced Voice Services (EVS); Performance characterization (3GPP TR 26.952)".
[i.9] STQ (12)40_32: "Proposal for correction factor when measuring receive part of super wide band
and full band headset terminals".
ETSI
9 ETSI TS 102 925 V1.2.1 (2018-10)
3 Definitions of terms and abbreviations
3.1 Terms
For the purposes of the present document, the following terms apply:
binaural listening: both ears are involved for the perception of sound
dichotic: relating to or involving the presentation of a stimulus to one ear that differs in some respect (as pitch,
loudness, frequency or energy) from a stimulus presented to the other ear
diotic: pertaining to or affecting both ears (same signal in both ears)
dual channel mode: audio mode, in which two audio channels with independent programme contents (e.g. bilingual)
are encoded within one audio bit stream
fullband: audio transmission bandwidth with a nominal pass-band wider than 50 Hz to 14 000 Hz, usually understood
to be 20 Hz to 20 000 Hz
stereo mode: audio mode in which two channels forming a stereo pair (left and right) are encoded within one bit stream
and for which the coding process is the same as for the Dual channel mode
super-wideband: audio transmission bandwidth with a nominal pass-band wider than 100 Hz to 7 000 Hz, usually
understood to be 50 Hz to 14 000 Hz
NOTE: Bandwidth definitions are adapted from Recommendation ITU-T P.10/G.100 [2].
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
AM-FM Amplitude Modulation - Frequency Modulation
CSS Composite Source Signal
DRP ear Drum Reference Point
DUT Device Under Test
EC Echo Cancellation
EL Echo Loss
EVS Enhanced Voice Services
FB FullBand
FFT Fast Fourier Transform
GAT Group Audio Terminal
G-MOS-LQO Overall Quality Mean Opinion Score, Listening Quality Objective, fullband
F
HATS Head And Torso Simulator
HFRP HandsFree Reference Point
IEC International Electrotechnical Comission
IP Internet Protocol
IPDV IP Packet Delay Variation
L Earcap Leakage
E
MCU Multiplexing Control Unit
MOS Mean Opinion Score
MRP Mouth Reference Point
MS Mid-sized Stereo
NIST National Institute of Standards and Technology
NLP Natural Language Processing
N-MOS-LQO Noise Quality Mean Opinion Score, Listening Quality Objective, fullband
F
PC Personal Computer
PDA Personal Digital Assistant
POI Point Of Interconnection
RLR Receive Loudness Rating
SLR Send Loudness Rating
ETSI
10 ETSI TS 102 925 V1.2.1 (2018-10)
S-MOS-LQO Speech Quality Mean Opinion Score, Listening Quality Objective, fullband
F
SWB Super-WideBand
TBD To Be Determined
TCL Terminal echo Coupling Loss
VAD Voice Activity Detector
4 Applications and Codec considerations
4.1 Applications
The following applications are within the scope of the present document:
• Speech and audio communication, including conferencing using high quality handsfree systems, for which
super-wideband/fullband coding can better reproduce the audio environment and provide an improved sound
quality, user's experience and audio immersion. These applications cover also GATs (Group Audio Terminals)
and teleconference systems such as "Telepresence".
• Bandwidth extension which may allow usage for some mixed content applications where wider bandwidth
could bring a significant added value for the customer (support of 14 kHz and 20 kHz bandwidth and
stereo/multichannel capability).
• Super-wideband enhancement coupled with stereo/multichannel to maximize the quality enhancement for the
customer when the terminal device can support this capability.
The send path can be characterized in two ways:
• The signal picked up by microphone(s) may combine speech, music and every type of environmental signal.
NOTE: For some applications (e.g. journalist reporting) the user should have the possibility to cancel the noise
environment or to transmit it without degradation.
• Direct insertion of any type of signal.
For receive path, the signal may combine the two following types:
• Communication signal such as described for send path.
• Signal coming from distributed applications (e.g. advertisement, music on hold, etc.).
4.2 Codec considerations
4.2.0 Introduction
As indicated in the scope only coders supporting conversational SWB and FB services are applicable to the present
document.
ETSI
11 ETSI TS 102 925 V1.2.1 (2018-10)
4.2.1 Super-wideband (SWB)
Table 4.2.1-1: List of super-wideband codecs covered by the present document
Coder Reference Speech Other signals Stereo Remark
ETSI TS 126 441 [25] X X Music (X) EVS codec.
Stereo supported
in a dual mono
configuration
Recommendation ITU-T G.722.1 [7] X X Music For low frame loss
Annex C
Recommendation ITU-T G.729.1 [8] X X background
Annex E (extension SWB) noise
(X) music
Recommendation ITU-T G.718 [9] X X Music
Annex B
Recommendation ITU-T G.711.1 [15] X X X (Annex F)
Annexes D and F
Recommendation ITU-T G.722 [20] X X X (Annex D)
Annexes B and D
Opus [30] X X X
When X is in brackets, it means that the coder is not optimized for this application.
The following codecs are recommended for super-wideband:
• Recommendation ITU-T G.722.1 [7] Low-complexity coding at 24 kbit/s and 32 kbit/s for handsfree operation
in systems with low frame loss. Annex C 14 kHz mode at 24 kbit/s, 32 kbit/s and 48 kbit/s.
- The algorithm is recommended for use in handsfree applications such as conferencing where there is a low
probability of frame loss. It may be used with speech or music inputs. The bit rate may be changed at any
20 ms frame boundary. New Annex C contains the description of a low-complexity extension mode to
G.722.1, which doubles the algorithm to permit 14 kHz audio bandwidth using a 32 kHz audio sample
rate, at 24 kbit/s, 32 kbit/s and 48 kbit/s.
- Annex C. This annex provides a description of the 14 kHz mode at 24 kbit/s, 32 kbit/s and 48 kbit/s for
this Recommendation.
• Recommendation ITU-T G.729.1 [8], Annex E (extension SWB for G.729.1 [8]).
- This annex provides the high-level description of the higher bit-rate extension of G.729 designed to
accommodate a wide range of input signals, such as speech, with background noise and even music.
• Recommendation ITU-T G.718 [9], Annex B Super-wideband scalable extension for Recommendation
ITU-T G.718 [9]). This annex describes a scalable Super-wideband (SWB, 50 - 14 000 Hz) speech and audio
coding algorithm operating from 36 to 48 kbit/s and interoperable with Recommendation ITU-T G.718 [9].
• Recommendation ITU-T G.711.1 [15], Annex D defines the Super-wideband extension.
- Annex F defines the Stereo embedded extension for Recommendation ITU-T G.711.1 [15].
- "Annex F is intended as a stereo extension to the G.711.1 wideband coding algorithm and its Super-
wideband Annex D. Compared to discrete two-channel (dual-mono) audio transmission, this stereo
extension G.711.1, Annex F saves valuable bandwidth for stereo transmission. It is specified to offer the
stereo capability while providing backward compatibility with the monaural core in an embedded
scalable way. The Annex provides very good quality for stereo speech contents (clean speech and noisy
speech with various stereo sound pickup systems: binaural, MS, etc.), and for most of the conditions it
provides significantly higher quality than low bitrate dual-mono. For some music contents, e.g. highly
reverberated and/or with diffuse sound, the algorithm may have some performance limitations and may
not perform as good as dual-mono codecs, however it achieves the quality of state-of-the-art parametric
stereo codecs."
ETSI
12 ETSI TS 102 925 V1.2.1 (2018-10)
• Recommendation ITU-T G.722 [20], Annex B defines the Super-wideband extension and Annex D defines the
Stereo embedded extension for Recommendation ITU-T G.722 [20].
- "Annex B describes a scalable Super-wideband (SWB, 50-14 000 Hz) speech and audio coding algorithm
operating at 64, 80 and 96 kbit/s. The Recommendation ITU-T G.722 Super-wideband extension codec is
interoperable with Recommendation ITU-T G.722. The output of the Recommendation ITU-T G.722
SWB coder has a bandwidth of 50-14000 Hz."
- "Annex D describes a stereo extension of the wideband codec G.722 and its Super-wideband extension,
G.722 Annex B. It is optimized for the transmission of stereo signals with limited additional bitrate,
while keeping full compatibility with both codecs. Annex D operates from 64 to 128 kbit/s with four
Super-wideband stereo bitrates at 80, 96, 112 and 128 kbit/s and two wideband stereo bitrates at 64 and
80 kbit/s".
• ETSI TS 126 441 [25]. The Enhanced Voice Services (EVS) codec consists of the multi-rate audio coder
optimized for operation with voice and music/mixed content signals, a source controlled rate scheme including
a voice/sound activity detector and a comfort noise generation system, and an error concealment mechanism to
combat the effects of transmission errors and lost packets.
- EVS is defined in ETSI TS 126 441 [25] and ETSI TR 126 952 [24]. The tests conducted on codec
implementations, e.g. [i.7] show that the requirements and test methods for SWB terminals as defined in
the present document apply.
- EVS is designed for packet-switched networks/Mobile VoIP and VoLTE is a key target application.
- The key features of EVS are Super-wideband speech (32 kHz sampling) with improved speech quality
and improved music performance.
4.2.2 Fullband (FB)
The following codecs are recommended for fullband:
• Recommendation ITU-T G.719 [10] Low-complexity, full-band audio coding for high-quality, conversational
applications.
- "Recommendation ITU-T G.719 [10] describes the G.719 coding algorithm for low-complexity full-band
conversational speech and audio, operating from 32 kbit/s up to 128 kbit/s".
- The encoder input and decoder output are sampled at 48 kHz. The codec enables full bandwidth, from
20 Hz to 20 kHz, encoding of speech, music and general audio content. The codec operates on 20-ms
frames and has an algorithmic delay of 40 ms.
NOTE: Recommendation ITU-T P.501 [1] Annex A specifies the use of the ISO base media file format as
container for the G.719 bitstream addresses non-conversational use cases of the codec (e.g. call waiting
music playback and recording of teleconferencing sessions, voice mail messages and online "jam"-
sessions).
• ETSI TS 126 441 [25]. The Enhanced Voice Services (EVS) codec consists of the multi-rate audio coder
optimized for operation with voice and music/mixed content signals, a source controlled rate scheme including
a voice/sound activity detector and a comfort noise generation system, and an error concealment mechanism to
combat the effects of transmission errors and lost packets.
- EVS is defined in ETSI TS 126 441 [25] and ETSI TR 126 952 [24]. The tests conducted on codec
implementations, e.g. [i.7] show that the requirements and test methods for FB terminals as defined in
this TS apply.
- EVS is designed for packet-switched networks/Mobile VoIP and VoLTE is a key target application.
- The key features of EVS are Fullband speech with improved speech quality and improved music
performance.
ETSI
13 ETSI TS 102 925 V1.2.1 (2018-10)
5 Test equipment and associated considerations
5.0 Introduction
The terminals within the scope of the present document are not only dedicated to speech communication but are also
mixing speech and audio contents and may implement stereo and multichannel transmissions. As a consequence there is
a need to define new parameters, such as:
• Loudness: Loudness Rating is determined only for speech or speech-like signals. Loudness may be calculated
over any type of signal (audio sequences, speech sequences and mix of these sequences). Moreover it is not
intended to define Loudness Rating algorithms for Super-wideband and fullband speech. To be consistent with
transmission planning, the loudness rating shall be determined for wideband calculation and loudness shall be
calculated. Clause 5.4.1.2 details the measurement principles.
• Binaural listening: The most of the test assessment methods and requirements for speech terminals are based
on monaural listening. Even if some of them (e.g. for Handsfree Loudness rating) are intended to take into
account binaural listening, the basic methods and requirements are only taking into account correction factors.
The plan is to adapt test methods to effective binaural listening.
As a consequence, the present document takes into account test arrangements that are defined for speech terminals or
for audio equipment.
5.1 Test Set-up
5.1.0 Introduction
Recommendation ITU-T P.58 [3] indicates:
"The artificial ears . support super-wideband as well as full-band applications. It should be noted that the
acoustical impedance of the artificial ears has some limitations in realistically simulating human ears".
"The artificial mouth supports super-wideband applications, however it should be noted that the directionality of the
artificial mouth is limited in its ability to simulate the human mouth in the super-wideband frequency range."
For terminals supporting SWB or FB a HATS (Head And Torso Simulator) should be used. For terminals supporting
SWB or FB in combination with Narrowband/Wideband functions a HATS (Head And Torso Simulator) shall be used
for parameters defined for limited bandwidth such as RLR and SLR.
For send path the HATS shall be used for super-wideband. Until the development of new systems with larger
bandwidth, send path measurements will be limited to super-wideband.
NOTE 1: Some HATS may provide a higher bandwidth. If a lab wants to apply the HATS for fullband testing, the
lab should check if the HATS used for t
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