PDA

View Full Version : Session over IP



Ian Alexander
05-11-2006, 08:43 PM
A friend sent an email this afternoon asking about a plugin (currently for PT) that basically replaces ISDN. It connects two PT computers over the Internet. The plug uses AAC encoding. Anybody heard of this? I had not. Anybody heard of this becoming available for other types of DAWs, like say SawStudio?

http://www.source-elements.com/

There's a quote on the site from our friend Rail. Maybe he knows if this will be released for other platforms or if competitors exist.

For most sessions, I think the bitrates are too low, but it sounds a lot better than a phone patch for monitoring remote talent.

Dave Labrecque
05-11-2006, 08:54 PM
A friend sent an email this afternoon asking about a plugin (currently for PT) that basically replaces ISDN. It connects two PT computers over the Internet. The plug uses AAC encoding. Anybody heard of this? I had not. Anybody heard of this becoming available for other types of DAWs, like say SawStudio?

http://www.source-elements.com/

There's a quote on the site from our friend Rail. Maybe he knows if this will be released for other platforms or if competitors exist.

For most sessions, I think the bitrates are too low, but it sounds a lot better than a phone patch for monitoring remote talent.

Ian,

I've heard about it, but that's all. Are you saying the bitrates aren't up to par with ISDN quality?

I did a session a year or two ago with a talent in L.A. who'd figured out how to send decent-quality audio over the Internet more-or-less live. He wouldn't divulge his method, but I was able to use Windows Media Player, type in an IP address, and away we went. I'd say it was comparable in quality to Layer 2 mono 128 ISDN. But there was like an 8 or 10 second delay, so I directed him over the phone and recorded him "in the background".

I'd think using the Internet is still a bit kludgy. But I'm open to breaking news... :)

Bob L
05-11-2006, 11:01 PM
For monitoring purposes, I would imagine a plugin could simply stream 128 or 192kbs mp3 format to a receiving system on the other end.

I am pretty amazed at the listening quality of SAWStudio Radio Network... its only a 128k bitstream.

Bob L

Ian Alexander
05-12-2006, 06:27 AM
Dave,

The Pro version offers a max bitrate of 320kbps stereo and 160kbps mono for $1495. The "talent" version offers a max bitrate of 160kbps stereo and 96kbps mono. By comparison, a single line ISDN connection provides 128kbps mono or stereo. For spots and video narration, 128 or 160 may work satisfactorily. For training projects that get delivered to the end-user via the web, though, the final files are subject to tremendous data compression. In that case, using any data compression at all on the original files is risky, IMO.

I'm surprised by the delay you report with your VO talent in LA. I've used MSN Messenger to audio conference with my cousin in Toronto. The delay is minimal and it sounds much better than a phone. Don't know that I'd record it, though.

I'm thinking it won't be long before non-proprietary solutions become available.

Naturally Digital
05-12-2006, 10:12 PM
I'm thinking it won't be long before non-proprietary solutions become available.Bring it on baby!

Veit Kenner
05-13-2006, 11:26 PM
I'm thinking it won't be long before non-proprietary solutions become available. Hi Ian,

I came across the VSTunnel plug-in at http://www.vstunnel.com/en/

It seems it could be used for a VO session as well, even if the community mostly speaks of musical collaboration.

The good thing is they offer a free version that is quite limited but good enough for testing.

Veit

Veit Kenner
05-14-2006, 02:17 AM
I came across the VSTunnel plug-in
... and there is another option:
http://www.digitalmusician.net/index.php?id=185&L=0

Dave Labrecque
05-14-2006, 05:32 PM
Dave,

The Pro version offers a max bitrate of 320kbps stereo and 160kbps mono for $1495. The "talent" version offers a max bitrate of 160kbps stereo and 96kbps mono. By comparison, a single line ISDN connection provides 128kbps mono or stereo. For spots and video narration, 128 or 160 may work satisfactorily. For training projects that get delivered to the end-user via the web, though, the final files are subject to tremendous data compression. In that case, using any data compression at all on the original files is risky, IMO.
Wow, that sounds pretty cool. Any idea what the latency is with this baby? I'd be surprised if it's anything close to ISDN, but I may very well could be wrong. ISDN still has the dedicated connection (and therefore bandwidth) that you can't get via the Internet. Or can you?

BTW, I don't think it's fair to say ISDN will give you 128 kbps mono or stereo. Isn't any mono over ISDN either 1 channel of 64 or simply 2 channels of 64, that, when properly synchronized, will allow for phase-coherent stereo?

How 'bout this: save half your long distance bill when doing Layer 2 sessions by only calling from/to one ISDN number. Perfect for VO: you get one channel of mono instead of two (and at half the long distance cost).

How 'bout this (corollary of the preceding): "Layer 2 mono 128" is actually stereo! Check it out. Why do they call it mono? I found this out once when the other end only sent me the voice on the left side because of a console routing error in their studio. :eek:


I'm surprised by the delay you report with your VO talent in LA. I've used MSN Messenger to audio conference with my cousin in Toronto. The delay is minimal and it sounds much better than a phone. Don't know that I'd record it, though.
Well, that's what I'm saying... it appears that to get good enough quality to record it (as good as ISDN to my ears), the protocols involved demand a huge amount of buffering, and therefore a large delay time. That's my best guess, anyway. :) That current Internet infrastructure is the limiting factor for bandwidth-on-demand.



I'm thinking it won't be long before non-proprietary solutions become available.

Ya know it's gotta be the future. :D

Ian Alexander
05-15-2006, 07:21 AM
Veit, thanks for the links. I will spend some time checking these out.

Dave, according to the manual for the MUSICAM Prima LT ISDN Codec: "When using 112 or 128 kb/s, the Prima LT delivers transparent, 20 kHz monaural or joint stereo or 10.2 kHz dual mono audio. MUSICAM Layer II at 128 kb/s, monaural, can stand up to extensive post-production and can be re-coded several times before artifacts become noticeable."

Maybe they have me bamboozled, but the display on the box is different when using mono, joint stereo, or dual mono. 128k also sounds better to me than 64k, even comparing 64k mono with 128k joint stereo.

I have learned that there is often confusion between displays when connecting Telos Zephyrs with other codecs, because the Zephyr still shows 64k (per ISDN B channel) even when connected at 128k. Perhaps there were cockpit problems on the other end of your left-only session. ISDN can be very user-unfriendly at times.

If you still have your Zephyr, let's do an experiment. I'll dial you on a single 64k B channel, then again on both B channels for 128k mono. I'll send the same audio so you can line em up on two SS tracks and A/B em.

Veit Kenner
05-15-2006, 10:44 AM
Veit, thanks for the links. I will spend some time checking these out.
Great. Let us know if you found some of them of value.

Veit

Dave Labrecque
05-15-2006, 04:28 PM
Veit, thanks for the links. I will spend some time checking these out.

Dave, according to the manual for the MUSICAM Prima LT ISDN Codec: "When using 112 or 128 kb/s, the Prima LT delivers transparent, 20 kHz monaural or joint stereo or 10.2 kHz dual mono audio. MUSICAM Layer II at 128 kb/s, monaural, can stand up to extensive post-production and can be re-coded several times before artifacts become noticeable."

Maybe they have me bamboozled, but the display on the box is different when using mono, joint stereo, or dual mono. 128k also sounds better to me than 64k, even comparing 64k mono with 128k joint stereo.

I have learned that there is often confusion between displays when connecting Telos Zephyrs with other codecs, because the Zephyr still shows 64k (per ISDN B channel) even when connected at 128k. Perhaps there were cockpit problems on the other end of your left-only session. ISDN can be very user-unfriendly at times.

If you still have your Zephyr, let's do an experiment. I'll dial you on a single 64k B channel, then again on both B channels for 128k mono. I'll send the same audio so you can line em up on two SS tracks and A/B em.

OK, pardner. Draw!

It makes sense, I think, that 128 kbps joint stereo would sound better than 64 kbps mono, since the former is taking advantage of any redundant data between the channels (which would be all of it in the case of VO!) when doing it's perceptual coding.

I'll PM you with my numbers and available times... :)