AAC-ELD Family

The low delay audio codecs for teleconferencing and OTT services

Already being widely used for professional and consumer communication applications, the AAC Low Delay codecs are state-of-the-art MPEG-4 audio codecs for maximum speech and audio quality at very low coding delay. By supporting the full audio bandwidth of 20 kHz, they are able to deliver Full-HD Voice audio quality to IP-communication applications and devices. These codecs can be regarded as a superset of each other, as they share the same coding core and each adds new coding tools. Software of the AAC-ELD codec family can be expected to be fully backward compatible.

The codecs of the AAC-ELD family enable various channel configurations from mono and stereo to a maximum of 48 channels.

Examples of AAC-ELD implementations include:

• Video-Telephony: Apple FaceTime
• Videoconferencing/Telepresence: Cisco, Tandberg, Polycom
• Operating systems: iOS, Android, Mac OS
• Broadcast contribution: Telos
• Standards: TIP, ETSI/Dect, OIPF, N/ACIP

NEW

Thanks to its new delay-reduced mode, AAC-ELD is now able to further reduce coding delay to 7.5 ms at a sampling rate of 48 kHz. This new mode is fully compliant to ISO/MPEG AAC standard. To learn more, please read our Technical paper “The AAC-ELD Family for High Quality Communication Services”  (p. 6).

ADOPTION AND APPLICATIONS OF THE AAC LOW DELAY CODECS

Over-the-Top Services and IP Video Telephony

Communication applications such as Skype often work with dedicated speech codecs. As these are optimized for speech signals but not for other signals such as music or arbitrary background noise, the delivered audio quality is limited. To overcome the limitations of speech codecs, Apple’s OTT peer-to-peer video telephony service FaceTime is based on the Full-HD Voice codec, AAC-ELD. As FaceTime is available on most Apple devices, such as iPhone, iPad, and Mac, the service can be used today on more than 200 million devices and is growing rapidly.

Video Conferencing and Telepresence

Video conferencing and telepresence services have been all-IP-based technologies for many years. In these markets, the user expectations of both video and audio quality are demanding.  As a result, Full-HD Voice has long been a default choice for these providers. Most companies are offering Full-HD Voice, and a majority of these products are based on the TIP standard, assuring interoperability between devices from different manufacturers. The TIP standard chose the AAC-LD as the only mandatory codec besides G.711, which is the legacy voice codec used in narrowband telephony.

Broadcast Contribution

Historically, live reports from remote locations have relied on switched ISDN network connections, where packet loss does not exist. ISDN connections are now being replaced by DSL Internet connections and mobile wireless data links. Neither of these connections is very reliable. This represents a major challenge for broadcast equipment manufacturers, as IP networks are best-effort networks that do not support guaranteed bit-rates and do not have constant delays. Because of the nature of broadcast contribution, this is a major issue. Low Delay codecs such as AAC-ELD can be a vital asset because they provide the low bit-rate capabilities and high-quality error concealment required to resolve these issues. As a result, AAC-ELD is already widely deployed in broadcast contribution devices such as the Telos Zephyr/IP and the Comrex Access.

TOMORROW’S COMMUNICATION APPLICATIONS

Telepresence at Home

Globalisation is having a profound effect on individuals’ business and private lives. The changing patterns of work and life are leading to a greater need for high-quality, easy-to-use personal communication solutions that allow people to keep in touch with friends, family and colleagues. The low delay audio codecs can provide the basis for comprehensive solutions integrated into broadband connected devices, including PCs, TVs, set-top boxes and mobile phones. Staying in touch should be as natural as having a conversation with all participants present in the same room. Fraunhofer’s low delay audio codecs allow service providers and hardware manufacturers to make this a reality.

The following operating systems natively support our low delay technologies:

• Fraunhofer FDK AAC Codec Library for Android™
• iOS (as an alternative: Apple iOS SDK)
• Mac OS X
  

Software licenses are available for:

• PC (Windows/Mac OS X/Linux)
• ARM
• MIPS
• Texas Instruments C6x, DaVinci, OMAP
• Analog Devices Blackfin

Further information about our development and design kits can be found on our "platforms" page.

If you are interested in licensing software from us please fill out the form below.

Patent licensing:

A patent licensing program is administered by Via Licensing Alliance. Please visit https://www.via-la.com/ for more information or contact contracts@via-la.com.

AAC-ELD is the superior, best-in-class codec for communication solutions requiring excellent audio quality at low bit-rates and with minimal latency.

AAC-ELD has been evaluated in several independent tests comparing it to codecs like G.718, G.719, G.722.1 Annex C (Siren 14), G.722.2 (AMR-WB), G.722, Silk, Speex and CELT. In these tests, AAC-ELD was found to deliver excellent quality at lower bit rates than any of its competitors.

Results can for example be found in:

Listening test results for 3G

Report on the verification test of MPEG-4 Enhanced Low Delay AAC

Background Information on audio quality and delay

Contemporary perceptual audio codecs such as MPEG Layer 3 (mp3) and MPEG-4 AAC offer good audio quality at bit rates down to 24 kbit/s. However, the algorithmic delay of the encoder/decoder chain exceeds 100 ms and is therefore not acceptable for two-way communication. By way of contrast, low delay speech coding schemes such as G.729 only provide satisfactory quality for speech signals and usually fail when used for complex audio signals like music.

The low delay audio codecs by Fraunhofer IIS are designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. They bridge the gap between speech coding schemes and high quality audio coding schemes.

Unlike common speech coders, the achieved coding quality of AAC-LD, AAC-ELD and AAC-ELDv2 scales up with bit rate, and transparent quality can be achieved. The audio quality of AAC-LD and AAC-ELD is superior to ITU-T G.722/G.722.1-C at identical bit rates. A structural comparison with the ITU-T pendant G.722.1-C (Siren 14) is discussed in the AES paper »MPEG-4 Enhanced Low Delay AAC - A new Standard for High Quality Communication« and is available on request.

Further, the AES paper »AAC-ELD v2 - The New State of the Art in High Quality Communication Audio Coding« (also available upon request) shows that despite doubled bit rate G.719 cannot outplay the audio quality delivered by AAC-ELDv2.