Full-HD Voice

AAC-ELD for Full-HD Voice

A widely adopted Full-HD Voice technology is the MPEG format Enhanced Low Delay AAC (AAC-ELD). AAC-ELD is a communication codec based on the commonplace music codec AAC. AAC-ELD provides CD-like audio quality for telephone calls from any audio source, including speech in all its variations, singing, music and ambient sounds. AAC-ELD supports full audio bandwidth up to 20 kHz and low coding delay down to 15 ms, which is crucial for real-time conversation. It is optimized for a bit-rate range of 24 kbit/s to 64 kbit/s per channel.


...enables Full-HD Voice

AAC-ELD delivers today’s best combination of low bit-rate, low coding delay and highest audio quality. It supports the full audio frequency range a human is able to hear and is not limited by the weaknesses of a speech codec. The result is a completely natural communication experience.

…is a proven technology

Millions of users are making their phone calls with AAC-ELD today thanks to its specification as the audio codec in numerous IP video telephony services, such as FaceTime. In addition, most major video conferencing and telepresence manufacturers rely on Full-HD Voice technologies in their products; Apple and Cisco are just a few of the globally-renowned companies to have selected AAC-ELD as their communication codec of choice.

…allows faster time to market

Fraunhofer IIS offers optimized AAC-ELD ‘ready-to-run’ implementations for all major platforms. This software greatly simplifies what would otherwise be highly time- and resource intensive implementation processes, and accelerates the product development stage.

…is easy to integrate

Most devices already run AAC codecs on their application processors. When using Fraunhofer’s optimized Core Design Kit software for AAC-ELD, in many cases, you may simply replace the existing codec library by a new one. This saves the service provider both time and money.

 …ensures a reliable service

In contrast to other communication codecs, AAC-ELD facilitates a consistently high quality of audio under bad network conditions. In addition, it adapts to changing transmission bandwidth and addresses transmission errors more efficiently than its competitors.