ISDN (VO's and remote colaboration)?

[tlester]

Hi all -

I understand how ISDN works and the technical aspects of this, but I've never done any real time collaboration. Can someone who does this regularly plese explain to me how you are doing this (from a practical perspective)? I.e. What kind of projects do you do this for? What's the quality? What's the work flow? Etc?

I don't have a pending project or whatever... I'm just curious. Realtime collaboration interests me A LOT. Just the idea of having players all over the globe working on an album together real time is pretty cool to me. I thought it was pretty cool when we had the built in rocket network thing (or whatever it was called), but this seems even more "real time" than that.

Thanks,
Thomas

[skyevox1]

I don't know how ISDN collaboration would work for real time musicians/players, but for Voiceover sessions it is perfect. I do VO from my ISDN studio - Zephyr & APT - almost daily. I am able to work to a music/jingle/effects track live, which greatly helps the performance. The engineer at the other end only has to offset the track by whatever the delay is in the round trip - about 11 frames each way on an APT Box - for playback. For the director in the studio, it is as if I am there. For VO, it is much better than tracking and sending via ftp, as you are not wearing two hats - trying to give the best performance and running a session at the same time.

For realtime collaboration for musicians, I think ISDN may not be the perfect tool, because of the delay/latency. The connection is full duplex, so one player could play against the others performance, but one would lead and the other would follow. The lead player wouldn't be able to be in sync with the following player, so would have to play, without hearing the other player. Kind of like tracking into an Mbox, with all the latency. But, if for example, a player was contributing a performance to an to an already built track, (drums - guitar - vocals) then ISDN would be great. The sending musician would still want to track in a DAW to be able to send the full bitrate files later, because all the ISDN codecs do lossy compression.

The Source-Connect plugin seems to have possibilities, but again delay, I think, would be a huge issue affecting any realtime performance.

Bottom line - for VO - the quality is great - a little less so for music (unless you're using a high end APT that require four ISDN lines) The workflow is the real benefit, as it puts both studios in the same space.

Bob

[spigots]

to further expound on the excellent post above,
apt makes an "uncompressed" codec. It requires 6 lines of isdn connection, however (can you say expensive ).

When i have tracked a live player or singer, only one end is "live". you figure out the delay before hand and send the "guide track" to the remote studio. The singer at the other end sings to what you are sending. Here's the trick; copy what your sending to another set of tracks (doing a rough 2-mix is a must) and delay your tracks to match the dely to and from the remote studio. You monitor this and the vocal. It works great. It would be impossible to do "live", so I don't know what folks do in that situation.

The easiest way to figure out the delay is to have the remote studio "foldback" what you are sending. Before the talent arrives, send a click or 2 pop and record the return from the remote studio. Then compare the difference of what you sent and what you got back. That's your delay.

Hope that made sense.

Frank

[Jimmix]

APT's codec is not actually uncompressed though it is nearly uncompressed. It works off of a delta/sigma difference information with a lookup table type scheme and is not perceptual like other mpeg based codecs. APT calls it ADPCM instead of PCM. They offer a new 24 bit codec called aptx. At the time of its introduction (at least 10 years ago) it was revolutionary and better than any other offering. AAC / mpeg 4 gives APT a run for its $$s though and is in fact more efficient based on its bit rates / quality which is also 24bit. Check out some examples. Each codec has its charater and it is often dependant on the program material it is working on. Besides a very quick encode and decode latency, one of the main advantages of the APT system is its ability to withstand multiple encode decode stages which is great for post production where for example radio commercials are often encoded with some mpeg variant for final delivery.

Also, the Source-Connect plug-in offers an excellent latency compensation system called RTS (remote transport sync). RTS automatically deals with and compensates for the latency between the two connected systems allowing for easy musical overdubs and ADR to picture over long distances.

Here is some good news: Source Elements announced a lower priced (about $400) version at NAMM. check it out:
http://www.createmagazine.com/news.c...go&NewsID=4285
here is a link about their RTS feature:
http://www.digidesign.com/news/detai...?story_id=4110

Jim