Today’s cellphones come with some fantastic features – touch screens, whizzy apps, you name it. The one thing, though, that they often fail to offer is the ability to make yourself heard clearly.
Videoconferencing, too, is often awkward because of time delay in the transmission. But a team at Fraunhofer IIS in Erlangen – the main inventor of the MP3 audio codec – says it’s come up with a way of drastically improving sound quality and reducing delay time.
Enhanced Low Delay Advanced Audio Coding, or AAC-ELD, gives a delay of only about 15 milliseconds. During this time, the algorithm manages to reduce the audio data to less than one-thirtieth of its original volume – without a major loss of sound quality, says the team.
It’s based on the existing AAC Low Delay algorithm, used by many existing video conferencing systems.
“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,” says team member Markus Schnell.
“We attempted to further minimize the area that is forward-looking and to only process current data. We did that until we found an optimum balance between quality and delay.”
The researchers say the codec can also improve speech transmission is also heard in mobile devices such as the iPhone 4 and the iPad 2. The developers have even created a special application that makes it possible to play games across the borders of cities or countries.
“Thanks to the optimized image and sound quality, there is the impression that game partners who are far apart from each other are not in front of screens, but actually sitting across from one another,” says team member Manfred Lutzky.
