Spatial Audio Conferencing

Spatial Audio Conferencing – for teleconferences with the "same-room-feeling"

With a truly amazing sound quality that creates the experience of "being there", Fraunhofer Spatial Audio Conferencing brings immersive audio to IP conferencing applications. The technology renders participants of a teleconference into a 3D soundstage and allows listening over stereo headsets. As voices are perceived from different directions and are reproduced in full audio bandwidth, conferencing becomes as pleasant and effective as having a conversation with someone in the same room.

More Productive Teleconferences

Hearing each participant from one distinct direction helps your brain to intuitively know who is talking. There is no need to think about who is talking and so you can focus on the spoken content. Additionally this makes a telco less tiring.

Natural Sound with Full-HD Voice

The voices of conference participants and all other surrounding sounds (including on-hold music and announcements) sound as natural and clear as known from "live talks" in the same room.

Scalable Operation in the Cloud

The technology is optimized for cloud-based deployments using Docker-containers. Its operation is therefore portable, scalable and cost-efficient.

AAC-ELD

By supporting the full audio bandwidth of 20 kHz, AAC-ELD delivers Full-HD Voice audio quality to IP-communication applications and devices for maximum speech and audio quality at very low coding delay. AAC-ELD is used by millions of FaceTime users every day and readily available on Android, iOS and OS X.

Binaural Mixing Library  

Most of the computational complexity for Spatial Audio Conferencing is handled by the server infrastructure, which uses a workload-optimized mixing library at its core. This library implements AAC-ELD decoding, binaural mixing for headphones, and AAC-ELD encoding for up to 50 participants on a single cloud-server. For third-party mono codecs, such as G.711, a PCM interface is provided to allow interoperability with legacy systems.

Low-Delay IP streaming system

To significantly improve the audio performance of existing networking equipment using its Audio Communication Engine, Fraunhofer IIS has created a new Low-Delay IP Streaming System to enhance connectivity. The system delivers optimized audio quality even under adverse network conditions such as packet loss and jitter. The component’s main features include:

• Sophisticated error concealment that guarantees good speech intelligibility at up to a 30 percent packet loss rate (understandable speech at up to 50 percent packet loss rate)
• Adaptive jitter buffer control to optimize the trade-off between buffering-delay and late-loss
• Log and trace files for remote analysis and debugging

A cooperation with Lindenbaum GmbH was established in order to bring Fraunhofer Spatial Conferencing to the market. Lindenbaum is a leader in audio and web conferencing platforms and services. Customers are telcos, conference service providers and enterprises in Europe and beyond.

Fraunhofer offers white label components for service providers and system integrators. The platform SDK includes libraries and extensive source code examples for easy integration.