TECHNICAL PAPER

Fraunhofer Institute for

Integrated Circuits IIS
Enhanced Voice Services (EVS) Codec
Management of the institute
Until now, telephone services have generally failed to offer a high-quality audio experience Prof. Dr.-Ing. Albert Heuberger
due to limitations such as very low audio bandwidth and poor performance on non- (executive)
speech signals. However, recent developments in speech and audio coding now promise a Dr.-Ing. Bernhard Grill
significant quality boost in conversational services, providing the full audio bandwidth for Am Wolfsmantel 33
a more natural experience, improved speech intelligibility and listening comfort. 91058 Erlangen
www.iis.fraunhofer.de
The recently standardized Enhanced Voice Service (EVS) codec is the first 3GPP commu-
nication codec providing super-wideband (SWB) audio bandwidth for improved speech Contact
quality already at 9.6 kbps. At the same time, the codec’s performance on other signals, Matthias Rose
like music or mixed content, is comparable to modern audio codecs. The key technology Phone +49 9131 776-6175
of the codec is a flexible switching scheme between specialized coding modes for speech matthias.rose@iis.fraunhofer.de
and music signals. The codec was jointly developed by the following companies, repre-
senting operators, terminal, infrastructure and chipset vendors, as well as leading speech Contact USA
and audio coding experts: Ericsson, Fraunhofer IIS, Huawei Technologies Co. Ltd, NOKIA Fraunhofer USA, Inc.
Corporation, NTT, NTT DOCOMO INC., ORANGE, Panasonic Corporation, Qualcomm Digital Media Technologies*
Incorporated, Samsung Electronics Co. Ltd, VoiceAge and ZTE Corporation. Phone +1 408 573 9900
codecs@dmt.fraunhofer.org
The objective of this paper is to provide a brief overview of the landscape of commu-
nication systems with special focus on the EVS codec. The main design constraints and Contact China
features are highlighted, while some brief technology insights are also incorporated. Toni Fiedler
Finally, listening test results conducted during the selection and characterization phase of china@iis.fraunhofer.de
the standardization process are presented and discussed.
Contact Japan
Fahim Nawabi
Phone: +81 90-4077-7609
fahim.nawabi@iis.fraunhofer.de

Contact Korea
Youngju Ju
Phone: +82 2 948 1291
youngju.ju@iis-extern.fraunhofer.de

* Fraunhofer USA Digital Media Technologies, a

division of Fraunhofer USA, Inc., promotes and
supports the products of Fraunhofer IIS in the
U. S.

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio
Communication Systems

When comparing the audio quality of a phone call with the sound of a movie on TV, the
muffled audio of the standard telephone conversation becomes evident to everyone. This is
mainly due to the limitation of the audio bandwidth in existing telephone systems.
Figure 1 illustrates the different audio bandwidth capabilities present in typical communica-
tion/broadcast systems and the human auditory system.

Figure 1: Audio bandwidth in broadcast

AUDIO QUALITY IN ENTERTAINMENT AND COMMUNICATION SYSTEMS and communication systems
20 Hz HUMANS CAN HEAR 20 kHz

ENTERTAINMENT EXPERIENCES

AM radio

FM radio

Streaming applications

TV broadcast

COMMUNICATION EXPERIENCES

POTS HD VOICE FULL-HD VOICE

Legacy systems *

HD Voice services with AMR-WB or G.722

Full-HD Voice services with EVS or AAC-ELD

0 Hz 50 Hz 300 Hz 3.4 kHz 7 kHz 14 kHz 20 kHz

*with AMR or G.711

Plain old telephone systems (POTS) provide narrow band (NB) audio signals, meaning
frequencies of up to 3.4 kHz of audio bandwidth only. HD Voice services deliver wideband
(WB) quality, where WB stands for an audio bandwidth of 7 kHz. Considering the
capability of the human auditory system, higher frequencies up to 20 kHz – relevant to
high-fidelity sound – are still missing. Therefore, HD Voice is further extended to Full-HD
Voice, including the quality levels of super-wideband (SWB) and full-band (FB). SWB stands
for an audio spectrum of 16 kHz, while FB contains all frequency components up to 20
kHz.

Landline telephone services today provide either NB or WB quality. The codecs used in
these systems are G.711 [1] or G.722 [2] both operating at 64 kbit/s. In the mobile world,
NB is the default quality level; however, an increasing number of WB services are now
emerging. The codecs used for NB and WB mobile services are AMR-NB [3] and AMR-WB
[4] both usually operating at bit rates around 12 kbit/s. Some mobile networks even allow
higher rates for AMR-WB, i.e. 23.85 kbit/s, although the quality improvement compared
to the default rates is rather limited. The codecs for mobile communications are highly
optimized for speech signals and, as a consequence, their capability for coding other
signals like music is not satisfying.

Specialized communication systems for telepresence or dedicated video conferencing

systems can already provide Full-HD Voice quality today. The de-facto codec standard for
such systems is AAC-(E)LD [5]. The codec operates on a wide range of bit rates, starting
from 24 kbit/s up to 64 kbit/s, while being able to transmit speech and music signals.

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 2 / 9
AAC-(E)LD is also utilized for so called over the top (OTT) services. Typical OTT applications
include Skype or Facetime, where the IP packet transmission is handled without the
network management of an operator.

The 3GPP EVS codec [6,7] overcomes the two major problems existing in mobile and
landline telephone systems, i.e. limited audio bandwidth and poor performance on
non-speech signals. At the same time, the codec is able to operate at typical bit rates for
mobile services. This paves the way for a new standard of user experience in communi-
cation quality, applicable to all kinds of networks, including landline, mobile and OTT.
In the following sections, this paper outlines the key technical aspects of the EVS codec
design which result in this significant step forward in service quality. It should be noted,
that many more aspects beside the audio coder may have a significant influence on the
end-to-end service user experience. These factors include the audio-front-end processing,
including echo cancellation, noise suppression, automatic gain control, wind noise filtering
and de-reverberation, or the network behavior causing delay jitter and packet loss.

Overview of the EVS codec

Design objectives

The EVS codec, as standardized by 3GPP in September 2014 [6], provides a wide range
of functionalities enabling unprecedented versatility and efficiency in communication. It
has been primarily designed for Voice over LTE (VoLTE) and fulfills all objectives defined by
3GPP, namely:
1. Enhanced quality and coding efficiency for narrowband (EVS-NB) and wideband
(EVS-WB) speech services;
2. Enhanced quality by the introduction of super-wideband (EVS-SWB) speech;
3. Enhanced quality for mixed content and music in conversational applications;
4. Robustness to packet loss and delay jitter;
5. Backward compatibility to the AMR-WB codec [20].

As pointed out previously, this paper focuses on aspects 2) and 3) of the design objectives.
For the sake of completeness, the quality enhancements for the legacy NB and WB servi-
ces in 1) are discussed later in this paper as well. Besides the improvements listed above,
EVS comes along with a full set of system functions required for communication systems
such as voice activity detection (VAD), discontinuous transmission (DTX), comfort noise
generation (CNG) or jitter buffer management (JBM). The codec operates at a wide span
of bit rates starting from 5.9 kbit/s up to 128 kbit/s, and as a consequence always provides
the optimal rate for each network. All definitions of the design constraints, developed
during the EVS standardization, are given in [6].

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 3 / 9
Technical Overview

Coding paradigms

In general, the world of audio coding can be divided into two paradigms:

• Speech coding: Approach to model the vocal tract of human beings

• Perceptual coding: Approach exploiting the limitations of the perception of human
auditory system

As described in [8], efficient speech coding schemes, such as AMR-NB and AMR-WB, have
typically three major components: (1) a short-term linear prediction (LP) filter modeling
the vocal tract; (2) a long-term prediction (LTP) filter, which models the periodicity in the
excitation signal from the vocal chords; and (3) an innovation codebook, for encoding the
non-predictive part of the speech signal.

Perceptual coding schemes, such as AAC [9], are based on three primary steps: (1) a time/
frequency conversion; (2) irrelevance reduction composed by a subsequent quantization
stage, in which the quantization error is controlled using information from a psychoa-
coustic model; and (3) redundancy reduction, featuring an encoding stage in which the
quantized spectral coefficients and corresponding side information are entropy-encoded
using code tables. This results in a source-controlled codec adapting to the input signal
statistics as well as to the characteristics of human perception.

In general, the speech coding approach offers best performance on pure, clean speech
signals at low bit rates, while the perceptual coding approach delivers better performance
on generic content, e.g. music, and provides up to perceptual transparent quality.

The first codec combining these two major coding approaches was the Unified Speech
and Audio Codec (USAC) [8]. USAC exceeds an algorithmic delay of more than 100ms,
which is not acceptable for bi-directional communication applications. However, motivated
by USAC’s excellent coding performance, the unified coding approach has been adopted
and further optimized to complement the challenging demands of the EVS codec.

Switched Speech/Audio Coding at Low Delay

The EVS codec is the first mobile communications codec to deploy content-driven, on-the-
fly switching between speech and audio compression at low algorithmic delay of 32 ms,
leading to significantly improved coding of generic content such as music signals.

The speech codec is an improved variant of Algebraic Code-Excited Linear Prediction

(ACELP), extended with specialized LP-based modes for different speech classes. For audio
coding, frequency domain (MDCT) coding is used. Special attention was paid to increase
the efficiency of MDCT based coding at low delay/low bitrates and on obtaining seamless
and reliable switching between the speech and the audio cores. Figure 2 depicts a high-
level block diagram of the EVS encoder and decoder.

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 4 / 9
 Figure 2: EVS codec structure

Super-wideband Coding and Beyond

EVS is able to provide SWB and even FB quality level and therefore overcomes the muffled
sound known from today’s telephony. Technically, the codec achieves this benefit by
utilizing bandwidth extensions. Depending on whether the speech or audio mode is
active, either a time-domain bandwidth extension (TBE) technology or an integrated
frequency domain solution is deployed. The latter provides several sub-modes, e.g.
harmonic model coding, which can cope with typical music signals. EVS is the first codec
providing differently optimized bandwidth extensions that are utilized and switched in a
source-controlled manner. Due to the dedicated content optimization, a very natural and
clean sound quality can be offered even at very low bit rates.

Performance Evaluation

Figure 3: 3GPP EVS characterization test

results for multi-bandwidth clean speech

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 5 / 9
 Figure 4: 3GPP EVS characterization test
results for multi-bandwidth mixed con-
tent and music

Extensive testing has been performed within 3GPP to verify the performance of the EVS
codec over a wide range of operating points and content types [11], including multi-
bandwidth tests conducted with the P.800 DCR method [12]. Figures 3 and 4 provide a
high-level impression of the quality (in DMOS score) for clean speech (English) and for
mixed content and music. The results include the quality levels narrow band, wideband
and super wideband at bit rates regarded as typical for mobile cellular services. The results
are discussed in the following:

• For EVS (operating in SWB mode), the clean speech quality level is already very high
for 9.6 kbit/s, outperforms AMR-WB 23.85 kbit/s significantly, and increases further
with bitrate. Starting from 13.2 kbit/s, the EVS-SWB clean speech quality already
approaches that of the “Direct Source” (original) quality.

• EVS (operating in SWB mode) outperforms AMR-WB even more significantly at

mixed & music content. Their grades differ in average more than 1.2 Mean Opinion
Score (MOS) at comparable bitrates. At 24.4 kbit/s, EVS mixed content & music
quality approaches that of the “Direct Source” (original) quality.

• Even when operated in WB mode (e.g. in devices or services only supporting WB

mode), EVS 9.6 kbit/s outperforms AMR-WB at 23.85 kbit/s. Furthermore it offers
much higher quality for clean speech and music when using an equivalent bit rate
(24.4 kbit/s).

• At first glance it may be surprising that AMR-WB cannot outperform AMR-NB for
mixed and music at same bit rate despite exploiting the double audio bandwidth.
Thanks to EVS, this weakness has been overcome.

• In the case of NB input signals, the EVS codec performs significantly better than
AMR-NB, for both clean speech and mixed and music content. This mode may be
useful in instances of interconnection to other NB networks such as landline.

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 6 / 9
It is well-known that test results and their interpretation vary with language and material
chosen. However, the EVS codec has been examined with regard to 10 languages, 6
different background noises and various music materials in the 3GPP Selection Phase,
showing excellent performance and improvement over earlier standards on a broad basis.
These results, combined with further extensive performance characterization of the EVS
codec, have been published in the 3GPP Technical Report (TR) 26.952 [11].

Applications

Since Long Term Evolution (LTE), the fourth generation of mobile network standards, has
been introduced, cellular phone networks are starting to switch to IP-based transmission.
LTE is based on the older, established GSM and UMTS standards, offering an all-IP
architecture and low latencies. It requires the deployment of all-IP voice services or Voice-
over-LTE (VoLTE) and in turn opens up the prospect of moving all voice services onto IP
networks, eventually phasing out the legacy-switched services based on GSM, UMTS and
CDMA networks.

With the help of Full-HD Voice technologies, service providers can shake off the limitations
of these legacy services, including very limited audio bandwidth and the use of speech-
centric codecs. Since VoLTE is providing Quality Of Service (QoS) in a managed network,
EVS heralds the opportunity to outperform OTT services such as Skype or Viber, not only
in audio quality but also in terms of robustness and service availability. Hence, mobile
operators will be able to regain lost ground with respect to voice minutes.

EVS, due to its outstanding error robustness [10], is also well-suited for usage in best
effort networks such as VoWifi (Voice over Wifi), and may also be available for 3G/circuit
switched systems in the future.

Conclusion

Various new features accompanied by unmatched speech and audio quality make the EVS
codec, the latest 3GPP codec for enhanced voice services, the most efficient and versatile
codec for high quality communication in any type of network, in particular cellular LTE and
Voice over WiFi networks. EVS opens up a completely new experience for users everywhe-
re, delivering an audio quality that is close to transparency even for mobile communication
services. The imminent introduction of the EVS codec will therefore bring lasting benefits
for both mobile operators and their customers.

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 7 / 9
References

[1] ITU-T Rec. G.711, “Pulse code modulation of (PCM) of voice frequencies”
[2] ITU-T Rec. G.722, “7 kHz audio-coding within 64 kbit/s”
[3] K. Järvinen. “Standardisation of the Adaptive Multirate Codec,” Proc. EUSIPCO, Sept.
2000.
[4] B. Bessette, et al., “The adaptive multi-rate wideband speech codec (AMR-WB),” IEEE
Trans. on Speech and Audio Processing, vol. 10, no. 8, pp. 620-636, Nov. 2002.
[5] M. Schnell, et al., “MPEG-4 Enhanced Low Delay AAC - a new standard for high
quality communication”, 125th AES Convention, Oct. 2008.
[6] S. Bruhn, et al., “Standardization of the new EVS Codec”, Proc. ICASSP, Apr. 2015.
[7] Martin Dietz et al., “Overview of the EVS Codec Architecture,” Proc. ICASSP, April
2015.
[8] M. Neuendorf, et al.: “Unified Speech and Audio Coding Scheme for high quality at
low bitrates”, Proc. ICASSP, Apr. 2009
[9] M. Bosi, et al., “ISO/IEC MPEG-2 Advanced Audio Coding”, paper 4382, 101st AES
Convention, Nov. 1996.
[10] V. Atti, et al., „Improved error resilience for VOLTE and VOIP with 3GPP EVS channel
aware coding“, Proc. ICASSP, Apr. 2015.
[11] 3GPP TR 26.952, “Universal Mobile Telecomm-unications System (UMTS); LTE; Codec
for Enhanced Voice Services (EVS); Performance characterization,”
http://www.3gpp.org/DynaReport/26952.htm
[12] ITU-T Rec. P.800, “Methods for Subjective Determination of Transmission Quality,“
Aug. 1996.

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 8 / 9
INFORMATION IN THIS DOCUMENT IS PROVIDED ‘AS IS‘ AND WITHOUT ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE.
INFORMATION IN THIS DOCUMENT IS OWNED AND COPYRIGHTED BY THE FRAUNHOFER- ABOUT FRAUNHOFER IIS
GESELLSCHAFT AND MAY BE CHANGED AND/OR UPDATED AT ANY TIME WITHOUT
FURTHER NOTICE. PERMISSION IS HEREBY NOT GRANTED FOR RESALE OR COMMERCIAL The Audio and Media Technologies di-
USE OF THIS SERVICE, IN WHOLE OR IN PART, NOR BY ITSELF OR INCORPORATED IN vision of Fraunhofer IIS has been an au-
ANOTHER PRODUCT. thority in its field for more than 25 years,
starting with the creation of mp3 and
Copyright © March 2017 Fraunhofer-Gesellschaft co-development of AAC formats. Today,
there are more than 10 billion licensed
products worldwide with Fraunhofer’s
media technologies, and over one billion
new products added every year. Besides
the global successes mp3 and AAC, the
Fraunhofer technologies that improve
consumers’ audio experiences include
Cingo® (spatial VR audio), Symphoria®
(automotive 3D audio), xHE-AAC (adapti-
ve streaming and digital radio), the 3GPP
EVS VoLTE codec (crystal clear telephone
calls), and the interactive and immersive
MPEG-H TV Audio System.

With the test plan for the Digital Cinema

Initiative and the recognized software
suite easyDCP, Fraunhofer IIS significantly
pushed the digitization of cinema. The
most recent technological achievement
for moving pictures is Realception®, a
tool for light-field data processing.

Fraunhofer IIS, based in Erlangen, Ger-

many, is one of 69 divisions of Fraunho-
fer-Gesellschaft, Europe’s largest applica-
tion-oriented research organization.

For more information, contact

amm-info@iis.fraunhofer.de, or visit
www.iis.fraunhofer.de/amm.

Audio and Media Technologies

Enhanced Voice Services (EVS) Codec www.iis.fraunhofer.de/audio 9 / 9