If YOUR groove VANISHES after Recording an OD... It AINT your ears... ADC ATE IT.....

[25ghosts]

Originally Posted by ronwasserman

Been messing with Studio One today. I think this may be 'the one' for some composing. I especially like that I can dup all my key commands so switching between PT and Stu 1 would be pretty easy. Nice that it loads up a project in about 15 seconds.......will report back as I SLOWLY learn this app.

Just wait until you find out just HOW intuitive it is. Crazy.

Good Luck !

[TLarvenz]

I downloaded it too. Looks good on first open. It crashed on first launch and then when I launched the second time, it said "not loading Battery 3 because it crashed last time." That's so much more helpful than any DAE error number I've ever gotten. Plain English. No searching the web. And it launched fine after that. So far so good!

Originally Posted by 25ghosts

Just wait until you find out just HOW intuitive it is. Crazy.

Good Luck !

[Mt.Everest]

Originally Posted by zakco

Why would you bother with micing up headphones when a simple loopback test (hardare out - hardware in) would be easier and more accurate?

Just for a more real world test I suppose. Technically, there shouldn't be any difference (other than the delay from headphone to mic capsule which is about 1/4" maybe) and it wouldn't matter if Im using the same setup for both tests-- ADC on and OFF, and checkboxes on and off.

Regardless, the point I was bringing up was why OP chose to UN-check the I/O boxes, when Digi has stated in the past that with 3rd party hardware, while its not perfect, its still better than NOT checking them.

Anyway, Ill have to catchup with the thread now to see where its at.

[Mt.Everest]

A bit off topic - although it does relate to ADC and Groove in the end -

I've asked this in quite a few posts since ADC has been around and I never seem to come out with something solid that stick in my head. So here goes:

SETUP
•• I work off of a Template. (One of several things in this list that I'm not willing to change)
•• In that template are a few Aux Tracks being fed by various groups of Audio tracks, such as Lead Vox audio trks 1 - 12 feeding LD VOX AUX, and Hook Vox audio tracks 1-24 feeding HK VOX AUX, etc.
••Each of these Sub-Aux's are sent to the Main 1-2 Output, or in some cases to my Master Aux which THEN feeds the Main 1-2 Output
••Each Sub-Aux has sends that feed various FX Aux Returns with Reverb Plugs, Delay, Chorus, etc.
This way, I can choose a Reverb send level for ALL my Lead Vox tracks once, and not have to put a send on each track. When I am tracking vocals for the first time for the given song, I prefer to record each take on its own track. So using an AUX for my overall " monitoring " level, FX levels, and sweetening is much easier than doing it track by track.
••Each Sub-Aux also has a few low-latency TDM plugs such as Compressor and EQ. (Again, not willing to change this method, even if my audio suffers)

So, before I even begin tracking, I already have a blank session template with about 150 - 175 samples total system delay.

I am currently running PT 9 HD and a Mac Pro Westmere.
I have the I/O checkboxes ON for compensate after record pass, etc.
I record with ADC ON. usually. When I think too much about it, I turn it OFF, as I'm still not exactly sure what happens working this way. (After years of discussion on this! Let the insults fly, by all means.. )

From what I remember in past threads, I am already starting off WRONG and my audio will never be placed correctly. It wont be MASSIVELY noticeable for vocals, but nevertheless, what gets sung is not what is heard back.

I have the Instrumental Mix playing back and feeding the Main 1-2 Out.
I record enable a vocal track, which feeds the Aux Vox Master which also feeds the Main 1-2 Out.
That track is automatically suspended from ADC when in record. But since it is passing thru an Aux, the talent is still hearing themselves delayed by X amount of samples. (Is that number the Total System Delay or just the delay on that specific Aux track?)

When I finish a take, take the track out of Record-Enable, what happens to the region I just recorded, if ADC was ON? How about ON but I/O checkboxes UN-checked?

Does it:
1. Physically move the region to match ADC's total delay?
2. Stay put, but plays back correctly via the genius smarts of ADC?
3. Does ADC take into account that the audio track Im recording on is being fed to an Aux, and NOT the Main 1-2 Out, and then adjust accordingly after recording? Or does it behave as if the audio track is being fed to the Main 1-2 Out ALWAYS?

I KNOW there is nothing that can be done for the delay incurred by the Aux it is being fed to, but shouldn't ADC behave differently based on WHERE my record-track is being fed to once I take the track out of record?

Since I refuse to output my vocal tracks all to Main 1-2 and have to deal with upwards of 60+ Sends since I'd have to put them on each track and also refuse to put 60 compressors on 60 tracks, Im sticking with recording my vocal takes thru an Aux.
Yes, I know I can have ONE track, and drag the region down after each take, or use playlists, and other methods. But for my workflow and how I keep notes on takes and how I comp later on, this is the way that works for me. I MUST record each take, whether 4 or 40, onto its own track. And each track while recording or comping must sound exactly the same, i.e. all having the same reverb, delay and compression levels on the Vox Aux. And each region name and number must match the track it was recorded on- i.e. Vox LD Vrs1Tk1 TRACK has to contain only regions that bear the same name, with the auto-numbering of PTs.

So, for any given time I am recording something (Fresh, Mid project, during mixing.. ) I disable high-latency plugs, no plugs on 2-mix obviously, and shoot for under 150 Samples Total System Delay which seems to be fine, for vocals at least.

The reason I write this is to ask again WHAT is actually happening to the audio regions using this method? Lets assume my Delay was 400 Samples. For one, yes, the talent will hear a delay, but how much? In the most basic form, I have a TSDelay of 150 samples, the Vox Aux has a delay of 3 samples. Overall, what is ADC doing while recording and/or ODing to already recorded tracks?

Lastly, am I making things WORSE by recording with ADC ON using my method? Should I turn it off temporarily, then back on after Im done with vocals for the day? How about the I/O check boxes? (Assume Im using Digi hardware too)

Thank you in advance! I know Ive asked these things before but I don't want work-arounds or alternate methods. I just want to know what ADC is doing in my Template Scenario, and if I'd be better off turning it off, given my method of recording vocals.

Thanks Again!
MT

[25ghosts]

Originally Posted by zakco

This thread is a non finisher. 25g was asked multiple times to do a simple loopback test as a repeatable example of this problem. He ignored that, which speaks volumes...

Zacko, didn't see this until now. Sorry.

The test I did was:

WHILE recording (Over Dubbing) I recorded the master two track. Then I bounced the result. So, in the end I had the PROOF-Two-Track (Taken WHILE recording) and the PROOF-Bounce (After OD was recorded I bounced the Mix.)

The recording done while recording the OD = In the Pocket
The Bounce done after Recording the OD = NOT in the Pocket.

I dont mean to be rude but I dont need to construct any further tests, to prove the point.
Which was - WHILE recording the OD, groove is IN the pocket. When listening BACK to the OD - Groove is OFF

I have reported it and I have dealt with it. I just DONT use ADC until ALL tracking has been done.

And I have been sailing just fine since doing that. SINCE doing this, I have not once had the following feeling:

GREAT while Recording - SUCKS when listened to !!!

[zakco]

Originally Posted by Mt.Everest

Just for a more real world test I suppose. Technically, there shouldn't be any difference (other than the delay from headphone to mic capsule which is about 1/4" maybe) and it wouldn't matter if Im using the same setup for both tests

The only reason I commented, was it just seemed more labour intensive and technically, (since I'm already nitpicking ) the signal is being passed through 2 transducers and therefore, potentially distorting the transients of the captured waveform. But...yes I get your point.

Z

[zakco]

Originally Posted by 25ghosts

I dont mean to be rude but I dont need to construct any further tests.

Well, you don't need to anything, but I can't for the life of me understand why you would not do a simple loopback test. This would be helpful in understanding the root of the problem. It would determine if bussing is part of the problem or not. This is NOT a small distinction...

I'm asking you to do this out of respect for those that have spent time and energy participating in this thread that you started. It takes minutes...less if you have a patchbay. Please try this and report back, thanks.

1. open a session with a ton of plugin induced latency, ADC on and the "compensate for input/output delays" checked.

2. Create an audio track with a click spike - actual audio track, not midi or VI (very important!!)

3. Bus this track ONLY to a physical output, no sends or mults.

4. Connect this output to a physical input

5. Create a new track with it's input set to the channel used in #4 and its output sent ONLY to your main outputs (assuming there are not plugins on the master).

6. Record the loopback audio

7. Zoom in and look at the waveform.

This is EXACTLY the same as hearing the cue mix, performing PERFECTLY (more than a human ever could). If your waveforms are in sync then the problem you're having must be rooted somewhere else. Most likely bussing.

I have never been able to make my system fail this test...

[25ghosts]

Originally Posted by zakco

Well, you don't need to anything, but I can't for the life of me understand why you would not do a simple loopback test. This would be helpful in understanding the root of the problem. It would determine if bussing is part of the problem or not. This is NOT a small distinction...

I'm asking you to do this out of respect for those that have spent time and energy participating in this thread that you started. It takes minutes...less if you have a patchbay. Please try this and report back, thanks.

Zacko,

this scenario has little to do with what I am/was experiencing.
I am NOT talking about a track that HAS been recorded or has a spike created to it.

I am talking about what happens DURING a recording. A recording which is being PLAYED by a performer. And how it sounds DURING and how it sounds when PLAYED back.

I DID read your scenario back then. But my opinion now and then was that it did not pertain to the problem I am/was having.

As you can clearly read off the thread; I was NOT complaining about an already recorded performance sliding off beat. But the difference between the groove while listening back to the recording WHILE it was recorded versus AFTER recording. Hard to do with a PT created spike.

[zakco]

Originally Posted by 25ghosts

Zacko,
I am NOT talking about a track that HAS been recorded or has a spike created to it....I am talking about what happens DURING a recording. A recording which is being PLAYED by a performer. And how it sounds DURING and how it sounds when PLAYED back.

Again, (sorry for being a broken record here) a click spike recaptured through loopback IS a perfect live performance, recorded and played back with a perfect, undeniable, timing reference.
It seems that you're completely missing what I'm saying. A loopback test IS the single most accurate way of determining whether or not what you're hearing while tracking is sample accurate to what gets captured and played back. It eliminates all human variables AND bussing related ADC weirdness (which we've all experienced I'm sure).

If you're really unwilling to try this, then I suppose I should let this horse die in peace, but I believe you're missing something important here. Or maybe I'm just an idiot.

Z

[25ghosts]

Originally Posted by zakco

Again, (sorry for being a broken record here) a click spike recaptured through loopback IS a perfect live performance, recorded and played back with a perfect, undeniable, timing reference.
It seems that you're completely missing what I'm saying. A loopback test IS the single most accurate way of determining whether or not what you're hearing while tracking is sample accurate to what gets captured and played back. It eliminates all human variables AND bussing related ADC weirdness (which we've all experienced I'm sure).

If you're really unwilling to try this, then I suppose I should let this horse die in peace, but I believe you're missing something important here. Or maybe I'm just an idiot.

Z

Missing point, perhaps.

BUT

what if there IS a difference between how ADC acts to LIVE input as opposed to something being played back from the timeline. If SUCH IS the case and I suspect it is, then your scenario would NOT reveal THAT.

AS you know... There IS a difference between MIDI being played back from a track AND midi being RECORDED to a track.

I DO understand and GRASP your scenario only IMO it does NOT apply to mine