Phil Wolff

Can Skype lead from telephony sound to theatrical audio?

July 3, 2005 03:28 PM

Topics: Developer Zone

"Can you hear me now?" is not the standard for sound quality. Anyone who's been to a modern cinema, sat in a high-end home theater, or played a sophisticated game knows there is more. After all, they give two Academy Awards for sound because sound makes movies more vivid, immersive, persuasive, emotional, and believable. The same is true for sound's effect on video gameplay, especially with team play.

As we move into an era when video and post-video experiences become part of everyday communication, what should our audio expectations include?

Don Tuite wrote a thoughtful essay for electronic design," Electronic Desgin logoAudio Codecs — The Entertainment-DSP Connection : Standard compression algorithms and proprietary post-processing code challenge DSP chipmakers to deliver more MIPS at fewer watts."

Until now we've been comparing audio quality of Skype and other Internet telephony tools to old fashioned telephones and mobile phones. Tuite suggests a path from telecom codecs (the software that turns sound into digital form, and back) to the highest quality audio DSPs. These deliver more channels of sound, spatial arrangement of those sounds in three dimensions, and more fidelity that ears can hear.

What's between today's high-end telecom quality and tomorrow's high-end entertainment quality? Technical barriers include heat, electrical power, processing demands, and chip size. Most of the high-end entertainment codec systems are proprietary and expensive.

Do you want conversations, movies, music, television, vlogcasts, and everything else we get through Skype to be exceptional? Let's set the highest expectations.

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Comments

Posted by: Rick at July 3, 2005 9:52 PM

Great thoughts. The distingushing factor between all of the codecs mentioned in the article and Skype's current codecs is the real-time nature of a voice conversation, as opposed to streaming audio, CD's, or even cellular phone calls. You cannot buffer it (in the broad sense), stream it, or store it and then play it as a delayed medium.

Voip is more problematic than just being real-time. IP is best effort packets that are sometimes dropped, received out of order, or delayed due to the IP network (i.e. not a continuous circuit call). Many factors determine the quality of Voip, more than can be mentioned here. IMO, a VoATM network would solve many of the network problems of IP, but that is not the current case.

Fortunately, Skype already has the best available technology on the market for Voip codecs. Global IP Sound is experts in this area with world reknown B. Klein as their Chief speech expert. When I heard my first Skype to Skype call, I was amazed at the sound... and it was no comparison to PSTN.

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