Audio on The Internet

Real-Time Transfer of Digital Audio Files

Unlike radio frequency broadcasts and other networks such as phone systems, the Internet was not designed to handle time-dependent data, such as audio material. Because the Internet spans a large geographical area and provides dependable stability, vast engineering resources have been spent to expand the possible capabilities. Applications such as NetPhone, an Internet telephony application, CU-SeeMe, a desktop videoconferencing application, and RealAudio, a broadcast-on-demand audio system, transfer audio/video information in, or near, real time. The bandwidth available is usually not adequate for high-quality sampling and bit rates; compromise and compression are standard. The systems do rival their traditional counterparts in functionality and usefulness.

The advantage of Internet-based telephony and videoconferencing is economic: for the same relatively low price one pays for Internet access, long-distance phone rates are bypassed. As some institutions already implement integrated voice and data services, leveraging those telecommunications lines for additional uses may be a prudent use of bandwidth.

Audio systems such as RealAudio use a buffering system to give the illusion of real-time transfer. The software will delay reproducing a transmission for up to two seconds in order to build a buffer of data, ensuring smooth reproduction in case of a delay in the link. Video systems will go even further by dropping video frames in order to relay undisturbed the often more vital audio element.

The RealAudio server now has the ability to broadcast live audio feeds. The function requires much more server capacity, and the buffering time may increase from 2 to about 10 seconds.

Next Section:
Transmission of Digital Audio Files: File Compression

Audio on The Internet