MP3-like approach to improve sound quality of telephones and video conferencing


May 30, 2011

Fraunhofer's Marc Gayer, Manfred Lutzky and Markus Schnell (L to R), developed AAC-ELD to improve the quality of communication systems (Image: Dirk Mahler)

Fraunhofer's Marc Gayer, Manfred Lutzky and Markus Schnell (L to R), developed AAC-ELD to improve the quality of communication systems (Image: Dirk Mahler)

Engineers from one of the main players responsible for the development of the MP3 codec, the Fraunhofer Institute for Integrated Circuits IIS, have developed a new audio coding technology that aims to provide telephone calls and video conferences with sound quality approaching that of direct communication, while at the same time cutting delay times that often sees both speakers talking over each other. The solution is called Enhanced Low Delay Advanced Audio Coding - or AAC-ELD - and researchers claim it results in long-distance communications that appear almost as if the participants are sitting across from each other.

The MP3 codec uses a lossy compression algorithm that essentially reduces the accuracy of certain parts of sound that are considered beyond the ability of most people to hear. The result is much smaller audio files that can be transmitted much faster while still sounding fairly faithful to the original audio. However, since the audio first needs to be encoded and then decoded at the listener's end, the Fraunhofer engineers faced a bit of a balancing act in developing something similar for interactive communications.

"The algorithm requires a certain amount of time to encode the data and to decode it again at the other end of the line. The process requires data that is still in the future, as it must wait for the data to arrive. This can result in a situation where interactive communication is very difficult," explains Markus Schnell.

After several years of trying to strike an optimum balance that gave the best possible quality with the shortest possible delay, the team decided on an algorithm that produced a delay of only about 15 milliseconds. The engineers say that is this timespan the algorithm manages to reduce the audio data to less than one-thirtieth of its original size without a major loss in sound quality.

The Fraunhofer team says the improved speech transmission offered by AAC-ELD could have applications not only for telephone calls, but also for video conferencing applications used on mobile devices and real-time chat during games.

About the Author
Darren Quick Darren's love of technology started in primary school with a Nintendo Game & Watch Donkey Kong (still functioning) and a Commodore VIC 20 computer (not still functioning). In high school he upgraded to a 286 PC, and he's been following Moore's law ever since. This love of technology continued through a number of university courses and crappy jobs until 2008, when his interests found a home at Gizmag. All articles by Darren Quick
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