Alright I found out why it disappears, but I'm really frustrated with the answer. I was running 88.2 sessions. Apparently you lose the ability to run ADAT past 48k in LE? Can someone confirm this? I think its time to go back to Logic if that's the case.

I still go to the DUC 'cause it has the largest concentration of PT users. AFAIK ADAT can only do 8 channels at 48k. At 96k you're down to 4 channels. I'm not sure that you could run 88k in HD either. Not with ADAT. I don't know about the limitation in LE. If you can afford Logic or if you have it already, go for it.

This might shed a bit of light.

http://recording.org/diy-pro-audio-forum/43353-48k-adat-%3E-96k-motu-what.html

I think what it boils down to is, ADATs only operate at 48K at it's highest rate. So there's no reason to receive or transfer at a higher rate.
I don't think it's a PT or logic thing. More likely its an ADAT protocol thing. So the work around was to at least get 4 tracks at 96K since it's the same amount of data down the lightpipe.

It is the same amount of data but because two light channels are used for one input you get double the info per input. That is why at 88.2/96k ADAT S/MUX only has four channels available at 88.2/96k and only 2 channel at 176.4/192k. Most keyboard optical outputs are not S/MUX but optical S/PDIF for instance and will only transmit two channels no matter the sample rate. But that is not your issue. The hardware design is your issue.

Pro Tools/AVID/Digi was and is known to limit bandwidth with all aspects of their LE programs and devices and even somewhat with the HD boxes. It does not surprise me at all. In fact at least through the 003 generation of devices AVID/Digidesign does NOT support S/MUX operation therefore the ADAT inputs are ONLY available at 44.1/48k sampling rate. This means you are stuck at the lower sample rates unless you use a different interface. Pro Tools isn't technically the issue but the device drivers and firmware for the 003.

From [="http://en.wikipedia.org/wiki/ADAT_Lightpipe"]wiki[/]="http://en.wikipedia…"]wiki[/]:

Lightpipe can carry eight channels of uncompressed digital audio at 24 bit [[url=http://="http://en.wikipedia…"]resolution[/]="http://en.wikipedia…"]resolution[/] at 48,000 [="http://en.wikipedia.org/wiki/Sampling_%28music%29"]samples[/]="http://en.wikipedia…"]samples[/] per second.
Although initially used for the transfer of digital audio between ADATs, the protocol was designed with future improvements in mind. All Lightpipe signals are transmitted at 24 bit resolution, no matter what the depth of the audio; information is contained within the Most Significant Bits and the rest of the bits remain a string of zeros. For example, if a 16 bit signal is sent via Lightpipe, the first sixteen bits contain the audio information while the other eight are simply occupied by zeros. The receiving device ignores information it cannot process. For example, a 20 bit signal going from a Type II ADAT to a Type I (which only operates at 16 bits) will simply ignore the bits below the sixteen MSBs.[[url=http://="http://en.wikipedia…"][3][/]="http://en.wikipedia…"][3][/]
Higher [[url=http://[/URL]="http://en.wikipedia…"]sample rates[/]="http://en.wikipedia…"]sample rates[/] can be used with a proportionately reduced number of channels, although the original ADAT machines did not support this. The Alesis HD24 recorder demultiplexes high sample rates across multiple ADAT channels. A signal with a sample rate of 96 kHz is thus effectively split up into two signals of 48 kHz each, maintaining audio compatibility with the original ADATs.

It's not that AVID/Digi actually limits the bandwidth. Anyone using the ADAT protocol has this limitation, as it's the limitation of the protocol itself.

It's my understanding, that the protocol standard was actually limited to 48kHz due to the fact that this was the foreseeable limit on converters... while the hardware limitations of the lightpipe devices hardware itself is on the order of 10-40Gbit/sec transfer rates.

I use ADAT at various sample rares all the time but all my devices support smux. Max, are you saying your 003 or other Digi box passes ADAT data to your DAW at higher rates? Over at DUC its stated over and over SMUX isn't supported.

I don't think Max is saying that. The Digi 00x family only supports ADAT protocol up to 48KHz sampling rates (i.e. no SMUX), although they support S/PDIF through the same optical optical connector to 96KHz.

ADAT protocol data is made up of 256-bit frames, each frame holding the payload data from 8 channels of 24 bits plus header and other control bits. The hardware constraints of Toslink connectors and plastic optical pipes for runs of a metre or two impose a bit rate limit of around 15Mb/sec. Transmitting 48000 ADAT frames per second results in an aggregate rate of 12.288 Mb/sec.

Boswell is correct...

The ADAT protocol is the culprit.

In the TDM system, the ADAT protocol is limited to 2 channels per pair of converter channels. Again, because the of the 256-bit frame.

Muxing two channels is about all that anyone's ever going to piece together. There's just too much data, for such a small data word to go any higher... and doubling the data word is the only method you have left... slipping half into one word, then the other half of the 2nd word of the 2nd channel. The timing after that, is just too tedious for too little gain.

I seriously doubt you would do any better than to then look at either MADI or AES as the real alternative... with preference to AES as there are more devices that actually can share AES than MADI... in the prosumer marketplace, anyway.

This ties back to another thread I've commented on, in regards to optical connectivity and the hardware letdown of it's alternative in copper.

Gotcha. The OP was trying to smux an 003 and couldn't figure out that Digi doesn't support it.

The other thread you're talking about is one of Chris's where he is trying to get the most stable digital format for high sampling. I agree with you that AES is probably Chris best bet.

ADAT is a decent enough format for my rig since I have enough i/o to get the channel count I need at 192 or 96 SMUX.

Not to drift too far OT, but isn't funny how the manufacturers touted 96 and 192, but then we got mp3's as the final 2 track?

While the benefit of 192 is pretty visible in low track count recordings of things like solo instrument and symphonic recordings, it just strikes me as ironic/goofy that so many "new" recordists want to track everything at the highest sample rate they can... then use poor/weak/cheap algorithms to generate empty3's for UToob.

24/44.1 is about as high as anyone really needs... unless you are specifically recording for video... and even then, there's very few broadcast networks that can even transmit at 96k... and even fewer consumers that can receive at 96... so even then, 48k is the broadcast standard for over the air HD... and even if they accept 5.1, the standard is still 48k.

With my live location/remote work, I always record at 44.1. There's already a high enough noise floor, that even running 24 bit, the average overall noise is running at around -24db. Most home recordists, (not a slam, btw) do not have a quiet enough environment, nor adequate drive space and data management to fully take advantage of the higher sample rates. They just get sucked into the hype of the sample rate wars and conform... then get really frustrated when things aren't as "great" as they thought it would be... and/or run out of storage and processing trying to deal with these higher sample rates that aren't really worth dealing with.

I'm absolutely with you there. I record everything at 24/44.1. As you say, the noise floor in my studio is not "pro" level. (Though back in the day there were probably lots of pro studios whose noise floor was buried under tape hiss.) I figure that recording at 96k would mostly give me a better image of the refrigerator motor upstairs in the kitchen and the traffic on Rt. 460.

Hmm. Sample rate and noise floor are two separate things.

We had some discussion of 44.1/96KHz recordings in [[url=http://[/URL]="http://recording.or…"]this[/]="http://recording.or…"]this[/] thread.

On the other hand higher sample rate means lower latency. There is some advantage when using Virtual Instruments. Other than that, 44.1 is good enough for me.

Boswell, post: 368163 wrote: Hmm. Sample rate and noise floor are two separate things.

We had some discussion of 44.1/96KHz recordings in [[url=http://[/URL]="http://recording.or…"]this[/]="http://recording.or…"]this[/] thread.

PREZACKLY!! e.g. too many lads believing hype, instead of knowing facts...

Max you make some good points. I don't hardly ever bother with 192. However, being one of those small track count recording ants at 30 paces sort of engineers I do like the sound/smoothness of 88.2k better for what I do.

Boswell, post: 368163 wrote: Hmm. Sample rate and noise floor are two separate things.

You are absolutely right, and I probably should have said this. But it is my impression that a relatively high noise floor of the ambient environment tends to obscure the advantages of higher sample rates.

Bob, you're the mathematician (hard for a Jarhead to spell) and Boswell is the circuit designer, but I thought higher sampling rates gave a resolution advantage for processing and editing-especially in the case of plugins etc. Now, in the case of good old rock n roll I completely see where the noise of the music itself sort of negates any advantage-perceived or otherwise-though I would think some of the folk or bluegrass stuff you do might benefit. We're still not talking 192k now just a double speed bump to 88.2/96k. I also will concede that wonderful and great recordings of all styles can be and are made at 44.1k every day.

If I had a way to edit the files easily I'd probably work in 1 bit anyway.

First, I should probably correct myself and say that I always record 24/44.1 when working in multitrack/overdub mode. Stereo pair and spot mics I do at 24/88.2. Just have not done much of that lately. I understand the benefits of the higher sample rates, but the audible benefits are hard to hear - even by experts in the best of circumstances. I haven't done much pure acoustic string music lately. Usually drums and electric guitar are in the mix. So - in my studio - that's not what I'd call "the best of circumstances." I'll throw [[url=http://[/URL]="http://www.josemanu…"]Jose's Album[/]="http://www.josemanu…"]Jose's Album[/] out as a sample of the results.

Now, I admit it may be getting close to time to rethink this. A big part of my reasoning for sticking with 44.1 is that it just made everything so much more stable. Last fall's computer upgrade may have rendered that judgement obsolete. I probably should do a test session at 88.2 and see how smoothly things go. If stability is really no longer an issue then even a small improvement in audio quality is worth it.