How is ADAT any further ahead in channels if I am wanting to stay at 88.2 then. The raydat is no further ahead and increasing my costs to get a work clock.

Michael Fossenkemper, post: 369157 wrote: Maybe I'm in my own world, but I can't remember the last time I used ADAT or SPDIF. I'm all AES and so is all my gear.

Yes, this is where I've been at for years. I'm with you on that Michael but now confused that I've been missing something all these years. Now we're getting somewhere. A debate to help audiokid! Yeah! Sleepless nights coming to an end. Help!

All my gear has BNC wordclock inputs or fed from the digital signal. I'm looking on the back of all my expensive digital mastering gear and it only has AES with BNC wordclock.

Here is what I need to connect the interface and converters to:

http://recording.org/attachments/mixdream_rear1500-jpg.936/

Notice that I am inserting all analog hardware via breakout cable via the MixDream's 1/8 and 9/16 inserts

Notice that the Direct Outs go into the AD DA 25pin connection of example ADI-8 QS 25pin analog dsubs. Clean and simple.

Notice Master Outs will go back to the DAW or further on to a Dangerous Master and then sent to a second Mastering DAW using the Lavry AD10

Attached files

gotcha. Nice piece of gear there.

hehe... I'm smiling but so confused... Please help audiokid choose the best interface for this hybrid DAW system?

For the sake of expanding mutual interest on how the real world connects to the DAW for various, I think its safe to say "departments" mobile recording, recording studios and mastering, I found this on Iron Maiden that sheds a bit more light on MADI for live.

I'm also thinking that ultra high end is still AES/EBU but the gap is closing in.

This may not seem related, but I think its very interesting watching the video on SPL's new [="http://recording.org/content/704-spl-neos.html"]NEOS[/]="http://recording.or…"]NEOS[/], a 120 volt ( +30 ), 24 channel analog summing mixer. The great divide is beginning between ITB and OTB analog summing studios. It will be a long time before I go back to believing ITB is the summing king.

To shake it up more... here is more interesting info worth reading:

[[url=http://="http://www.acousenc…"]TECHNOLOGY[/]="http://www.acousenc…"]TECHNOLOGY[/]

Be sure to check out the Album of the week

audiokid, post: 369396 wrote: I'm also thinking that ultra high end is still AES/EBU but the gap is closing in.

Maybe about connectors but in my opinion Bits are Bits, so sound should not be affected unless there is Jitter or errors from bad wires/connectors.

Out of interest, what is the signal quality difference between AES/EBU, the optical format or SPDIF at the same SR / BR?

No, I rather don't ask... Too complex... It'll end in jitter and WC topics...
;-)

So from an interface purchasing decision, it all comes down to what way I want to connect my converters. Options, limitation and expanding plus software supported by the interface have nothing to do with its sound quality? Bits are bits and the clock is very important once we start expanding.

Taking that into consideration, it goes right back at choosing a converter that connects cleanly to my hardware and anything beyond 96k is really not something I will use.

I've baked this topic to death... Thank you everyone, you've all so helpful!!

So, the ADI-8 QS is a better choice than Lynx being more focused on AEA/EBU. Converters between the two may be similar and the added features for the RME is what I'm paying more for.

Big K, post: 369408 wrote: Out of interest, what is the signal quality difference between AES/EBU, the optical format or SPDIF at the same SR / BR?

Hi Rainer,

This is why I' ve been asking this, and unsure, but, I believe AES/EBU is best for shorter runs and MADI is best for long. I'm looking forward to your question being discussed though,

Big K, post: 369408 wrote: Out of interest, what is the signal quality difference between AES/EBU, the optical format or SPDIF at the same SR / BR?

I think this could help: [[url=http://[/URL]="http://www.hardware…"]S/PDIF - HwB[/]="http://www.hardware…"]S/PDIF - HwB[/] .

MADI makes more sense but it doesn't convince me its the best for my hybrid system and the tools I current am using and planning on buying. I hope I'm not wrong.

Big K, post: 369408 wrote: Out of interest, what is the signal quality difference between AES/EBU, the optical format or SPDIF at the same SR / BR?

No, I rather don't ask... Too complex... It'll end in jitter and WC topics...
;-)

We have the WC Jitter covered very nicely and most likely better than anywhere else on the planet with deep thanks to McEase :
http://recording.org/diy-pro-audio-forum/45012-what-clock-jitter.html

I'm posting out of sync with you both but after digesting the link Richard posted (thank you!) I'm still a child at this but gaining.

I can see (correct me please) how we need two separate systems for a few reasons if we really want to tune into a tracking and "overdubbing" system vs a proficient mastering system where little concerns over latency matters. Thus, why Richard can use Firewire and why I can't. Its also why I'm leaning towards the ADI-8 QS for a balance between efficiency and sound quality. How do I know this, I don't, but I have a feeling I'm right?

I thought about investing in 16 Lavry Blues for my DAC but I have a feeling ( tell me is I am wrong) there would be latency that I would notice. Again, I'm not even close to understanding the science here but the better the clocking, the more latency is in the specs? Am I off on this?

Why does one converter have more latency than the other?

Chris, it would be hard to go wrong with either AES or MADI as long as you have converter/connections on both ends of the cable to fit. You are correct thinking that on very long runs MADI has an advantage but on short runs it is a crapshoot near as I can tell. The quality of the wire/light pipe and the quality of the connectors are going to be far more important.

One of the things that has been brought to my mind lately from Soap's issues with the RME is cable strain relief. All of us that have had touring rigs or live PA rigs generally had means of relieving strain on all those multi channel snake harness both at the splitter box and at the console. How many of us actually do that at home in the studio? I'd say not as often as we should. I know in the Corps I even had the studio wired with strain relief and screw down D-Subs into the patch bays to alleviate things but since I've been a Civilian I am not nearly so careful. I guess what I'm saying is that the best gear in the world can use the best wire in the world but if we don't take it easy on the jacks and plugs quality is degraded. That is especially true now that all these boxes have PCB based I/O instead of chassis mounted metal connectors with bolts and point to point wiring. Stick a firewire cable or usb cable in to a jack sometime and see if it wiggles. I bet it does.

Better clocking is more accurate digital transmission. Any latency introduced is really at the computer end or any device in the middle that processes the audio. A star clocking array is the way to overcome any latency from the WC due to daisy chain.

Re cable strain: Man did you ever hit the nail on the head there. When I was recording those choirs last month I was freaking over how the firewire cable fitted into the laptop FW. What a total toy I though. Its definitely getting pitched as soon as I can afford a better laptop interface.

For my rig, I'm going AES/EBU and very short cable. 8 ft for the AES and 3ft for the dsub snakes. . I'm almost ready to make the move.

I use the SKB Studio Flyer. The firewire cable comes up through the whole and loops around once and then plugs into the Express Card providing it's own support. It's not perfect strain relief but better than hanging off the card pulling it down.

Wow, what a cool deal John! I've got to get that. [[url=http://[/URL]="http://www.bestbuy…"]Other Image Views[/]="http://www.bestbuy…"]Other Image Views[/]

$269.00

Yup. That's the one. I have the laptop on top of course and the four spaces are Furman conditioner (soon to be replaced by Furman UPS), True Systems Precision 8, Glyph GT050Q & GT051 combo, FF800. The True P8 has dual balanced output (8 TRS and D25). The TRS gozoutas are cabled into the FF800 and the DB25 goes to the Alesis HD24XR when I don't use ADAT to the Alesis. AND IT ROLLS!!!! :-)

Most word clocks from dedicated clocking units and those generated by decent cards, I/Os and converters ( if they posess this feature) are quite stabile and usable these days.
External dediacted WD generators ( like Nanoclocks ) are being used if a number of devices need to be clocked together for synch and/or the units are farther apart or other rooms/directions.
Wordclock signal travels with SoL and from that it can be done daisy chain style without introducing any additional lateny to any I/O devices.
Due to problems with the correct 75 Ohm termination settings, as well as the usual suspects ( Power supply and ground plane noise, interlogic interference,
cable interference, increased noise and clock skew hrough signal loops) a starshaped cabeling is adviceable. A dedicated WD units is not better than the WC comming
from a decent card like RME's, BUT a WC generator is much more flexible, has much more outs, can be used as Generator and as a Distributor ( set your clock on DAW or
PC and the distributor switches all devices to the same synchronized clock value). The signal voltage from a WC units is more stabile when a number of units is connected, thus the
signal will not go below a critical level. Any additional device, maybe hooked up just temporarily, can be synched more easily... many more.

I would recommend 96kHz rather than 192kHz. It’s hard enough for clocks to keep stable at 96kHz. Even the best designs can have difficulty. Additionally, cabling and interfacing can add another potential for error to an already struggling system. IMO the theoretical sonic improvement of 192kHz causes more concern than the benefits of a more stable 96kHz system.

Solid advice, thank you all! I'm at 88.2 most of the time and I see no point changing. I bet we'll see 384 in not too long.

audiokid, post: 370283 wrote: I'm at 88.2 most of the time and I see no point changing. I bet we'll see 384 in not too long.

Here you go: [[url=http://[/URL]="http://www.merging…"]Merging Technologies[/]="http://www.merging…"]Merging Technologies[/]

Very interesting company, thanks Richard. Are you using any of their software or hardware?

audiokid, post: 370406 wrote: Very interesting company, thanks Richard. Are you using any of their software or hardware?

No, I don't. I heard about them (company) but that's about it.

They seem to have a lot covered, what looks to be the design of pretty high end converters to broadcasting and video, mastering, to mass track count software for serious recording and mixing. The mixer strips looks really nice.
I spent an hour looking around there yesterday and would love to hear more about them.

The 4 point editing looks cool and working with video was very impressive. I'm not a mastering engineer so I quickly moved on to the recording software.

My first impression on configuring a session was a bit busy for my attention but I'm guessing like all new things, confusion soon takes on simplicity the more you become familiar. The video on recording focused on importing tracks to much where I would have liked to have see how it handled a 24 track session at first glance. I'm definitely going to pay attention to this company.

Routing looks to be amazing and definitely right on for hybrid studio's.

384, wow, can't imagine!

Do you use Sequoia?

I am using Samplitude for at least the last 8 years. I only do Mastering so I don't need much. Simpler the better for me. I would hate to start over with other programs unless this would make it sounds better.

Richard

Sonically, I can't imagine better than Samplitude/Sequoia. I'm excited for version 12 slated for this summer.