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Is an analog mixer cue mix better than Pro Tools HDX cue mix?
Old 6 days ago
  #1
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Is an analog mixer cue mix better than Pro Tools HDX cue mix?

I'm just curious because I have an HDX system however I remember hearing Fab Dupont talk about how a cue mix from "copper"(analog mixer) is just way better than a cue mix from a DSP powered system.

I never tracked vocals with an analog mixer cue mix but I'm just curious if in fact in general people that have experience with both favor the analog mixer cue mix.

Is it any different? Fab Dupont claims even HDX systems will have some latency. He says he uses his Neve for tracking and uses it monitoring as well and goes on to say "players play better because they fell better". From personal experience I will say back when I had an Mbox 3 Pro that latency around 64 buffers wasnt really an issue however the sound was a bit weird. It sounded like it had a the tinniest amount of reverb and when I used the Mbox 3 Pro's internal mixer it actually sounded really good compared to Pro Tools round trip sound. In that situation I have to admit Mbox's internal mixer was way better that Pro Tools mixer but I can do both with no problems.

What experiences do you guys have?
Old 6 days ago
  #2
Absolutely yes. No latency is better than any latency, no matter how small. Excellent musicians can feel the difference between none and “only a little”.
Old 6 days ago
  #3
That's why interfaces that don't have built-in direct monitoring are a no go.
Old 6 days ago
  #4
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Sound travels at very roughly 1.1 ft/millisecond.

At 44.1 khz, the round trip latency through HDX is 1.7 milliseconds.

So if you mic a kick drum and feed the signal through HDX to the drummer's cans, he will hear the signal in his cans before it arrives through the air. He will hear his snare right about in time with the air signal.

It's when you triple/quadruple those numbers that things start getting weird. Such as monitoring through a native CPU buffer.
Old 6 days ago
  #5
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Quote:
Originally Posted by bgrotto View Post
Absolutely yes. No latency is better than any latency, no matter how small. Excellent musicians can feel the difference between none and “only a little”.
There's no such thing as "no latency", and the amount of latency in an HDX rig is well below the 5 to 10ms of necessary for the most acute musicians to detect.
Old 6 days ago
  #6
Nashville musicians are unbelievably picky when it comes to latency so I had to drop my old HD Native system.

When I got HDX, even the top guys couldn't detect it. It's been smooth sailing ever since.
Old 6 days ago
  #7
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Brent Hahn's Avatar
 

I'm so used to hdx and tdm latency that a straight wire sounds a bit weird to me.
Old 6 days ago
  #8
After a couple of decades of real time monitoring, I find it unsettling to use the 'near zero latency' monitoring built into my interface when I'm doing direct in with guitar.

It would be a blessing to be able to do so at times, but I just find it very... uncomfortable. Not undoable (that would be playing into amp sims). But certainly uncomfortable. It feels like my fingers are 'sticky'... Despite having a crappy sense of rhythm (to my thinking), I am nonetheless acutely sensitive to very minor time differences.

(There was a time in the early part of the last decade when the latest version of my DAW claimed to have perfect time alignment between previously and newly recorded tracks, as tested by a loopback. I was naive enough to believe it but one day not long after adopting the update, I tracked a bongo part. As soon as I played it back, I could tell it was all off -- in addition to my own terrible rhythm. Not part off... all off, all the way through, every strike. When I did a loopback test I found newly recorded tracks were between 7.7 or so milliseconds behind previously recorded ref tracks. The DAW didn't have a way of adjusting for it yet, so I had to calculate the correct number of MIDI ticks to nudge, it was a total PITA, but it was better than having tracking that was 7 ms off. I tell this story to underline my sensitivity to time issues. Despite my poor rhythm. )

I totally get the sound-in-free-air math but I'm reporting my experience, here. And I don't dig the sensation of hearing my live part delayed in the monitor/cue mix. It doesn't feel right. (And if the delay is long enough, it doesn't sound right.)

Last edited by theblue1; 6 days ago at 02:25 AM..
Old 5 days ago
  #9
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Quote:
Originally Posted by theblue1 View Post
After a couple of decades of real time monitoring, I find it unsettling to use the 'near zero latency' monitoring built into my interface when I'm doing direct in with guitar.

It would be a blessing to be able to do so at times, but I just find it very... uncomfortable. Not undoable (that would be playing into amp sims). But certainly uncomfortable. It feels like my fingers are 'sticky'... Despite having a crappy sense of rhythm (to my thinking), I am nonetheless acutely sensitive to very minor time differences.

(There was a time in the early part of the last decade when the latest version of my DAW claimed to have perfect time alignment between previously and newly recorded tracks, as tested by a loopback. I was naive enough to believe it but one day not long after adopting the update, I tracked a bongo part. As soon as I played it back, I could tell it was all off -- in addition to my own terrible rhythm. Not part off... all off, all the way through, every strike. When I did a loopback test I found newly recorded tracks were between 7.7 or so milliseconds behind previously recorded ref tracks. The DAW didn't have a way of adjusting for it yet, so I had to calculate the correct number of MIDI ticks to nudge, it was a total PITA, but it was better than having tracking that was 7 ms off. I tell this story to underline my sensitivity to time issues. Despite my poor rhythm. )

I totally get the sound-in-free-air math but I'm reporting my experience, here. And I don't dig the sensation of hearing my live part delayed in the monitor/cue mix. It doesn't feel right. (And if the delay is long enough, it doesn't sound right.)
That's a different issue - somewhere along the line, you had a problem where your daw was printing with latency - above the level where most good musicians begin to hear it ( 5-10ms).

The OP is talking about real-time monitoring - HDX's real-time monitoring should be competitive with any electronic monitoring set up - analog or otherwise.

There's latency in everything we do, our brains compensate. If you're a pianist, you're familiar with latency, a LOT of it. This is a well researched field with lots of brain studies on how humans react to latency, I think the OP is worrying about something he doesn't need to ... given he has a HDX system and it's working well, he'd have one of the most "latency free" setups available.
Old 5 days ago
  #10
Quote:
Originally Posted by Sharp11 View Post
That's a different issue - somewhere along the line, you had a problem where your daw was printing with latency - above the level where most good musicians begin to hear it ( 5-10ms).

The OP is talking about real-time monitoring - HDX's real-time monitoring should be competitive with any electronic monitoring set up - analog or otherwise.

There's latency in everything we do, our brains compensate. If you're a pianist, you're familiar with latency, a LOT of it. This is a well researched field with lots of brain studies on how humans react to latency, I think the OP is worrying about something he doesn't need to ... given he has a HDX system and it's working well, he'd have one of the most "latency free" setups available.
I was citing the earlier problem -- and put that paragraph in parentheses to offset it from the rest of the post -- to demonstrate that as little an uncorrected offset as 8 ms can make a performance unusable -- and so unusable, the timing mismatch was instantly noticeable simply from listening to the track in context with other tracks.

Now, for sure 8 ms is not 2 ms (a typical turnaround for post-AD/DA direct monitoring) -- but it's my position that since it's entirely possible for us to monitor with a tiny fraction of that latency (at least using analog wired or using now gray-legality old analog wireless systems), that is the only acceptable baseline. That's how the monitoring on my tape decks worked. That's how I insist my tracking cue and monitoring work. If others want to adjust their head to delayed input monitoring, that's their business. Maybe others can't hear or feel an offset of 2 ms while monitoring input but I can. And it makes me quite uncomfortable. But since it's not necessary for my practice, I refuse to work like that.

As a keyboard dabbler, I get it that we can adjust for latencies in our performance gear, something synthesists come to do somewhat on the fly because of different envelopes (not to mention the nasty matrix of latency involved in many MIDI daw setups).

But I've spent nearly five decades playing guitars while listening to them in real time -- and nearly forty years overdubbing via analog monitoring into headphones.

I guess my sense of rhythm just isn't good enough to comfortably play well while compensating for an 'arbitrary' delay in what I'm hearing from my guitar.
Old 5 days ago
  #11
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Muser's Avatar
zero latency is probably ideal because it at least allows you to add more delay under a controlled condition. baring that, getting as close to zero as possible is probably second best for similar reasons. depending on what you are playing as a sound source, can also have additional issues especially if the device in question is a sample based device.

it could be useful in sample based technology that with higher velocities, that the sample start moved closer to a given start position, from a given onset position. this is usually currently, not the case however. this video contains a good example of why that could be desirable.

Old 5 days ago
  #12
Quote:
Originally Posted by Muser View Post
zero latency is probably ideal because it at least allows you to add more delay under a controlled condition. baring that, getting as close to zero as possible is probably second best for similar reasons. depending on what you are playing as a sound source, can also have additional issues especially if the device in question is a sample based device.

it could be useful in sample based technology that with higher velocities, that the sample start moved closer to a given start position, from a given onset position. this is usually currently, not the case however. this video contains a good example of why that could be desirable.

Having worked considerably with synths and to a slightly lesser degree with samplers before I ever worked with a DAW (where I could see the sample envelopes on screen), I was already well aware that the musical beginning of a sampled note seldom lined up with the beginning of the sample itself, typically requiring the performer to play 'ahead' in order to get that musical start time in the right place. (And I was familiar with the phenom from working with certain synth patchings whose envelope called out for playing ahead, as well.)

It's for that reason that when I'm editing timing of anything, synths, samples, or tracked instruments, I don't rely on visuals for anything but initial placement, drilling in the fine placement by ear alone.
Old 5 days ago
  #13
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Muser's Avatar
Quote:
Originally Posted by theblue1 View Post
Having worked considerably with synths and to a slightly lesser degree with samplers before I ever worked with a DAW (where I could see the sample envelopes on screen), I was already well aware that the musical beginning of a sampled note seldom lined up with the beginning of the sample itself, typically requiring the performer to play 'ahead' in order to get that musical start time in the right place. (And I was familiar with the phenom from working with certain synth patchings whose envelope called out for playing ahead, as well.)

It's for that reason that when I'm editing timing of anything, synths, samples, or tracked instruments, I don't rely on visuals for anything but initial placement, drilling in the fine placement by ear alone.
the primary difficulty is usually making sure you have a condition where everything comes back as you heard it / played it, in the first instance. if you assume regardless of factors related to the instrument, you played something perfectly, then it should be perfectly reproduced when played back. it's the factors which can make that vary, which can be difficult to properly factor out as a variable. depending on what you're using of course.
Old 5 days ago
  #14
Quote:
Originally Posted by Muser View Post
the primary difficulty is usually making sure you have a condition where everything comes back as you heard it / played it, in the first instance. if you assume regardless of factors related to the instrument, you played something perfectly, then it should be perfectly reproduced when played back. it's the factors which can make that vary, which can be difficult to properly factor out as a variable. depending on what you're using of course.
You bet. Having moved to a DAW from a hybrid 16 track MIDI/ADAT tape rig back in 1996, I've seen DAW use go from ultra-tweaky, nobody-breathe precariousness to relatively reasonable dependability and ease of use and even setup. Emphasis on relative.

That said, I nonetheless do a ping loopback test typically once or twice a week when I'm working to make sure things are still on the square. (Updates these days are frequent; occasionally that's meant my settings getting rejiggered. My DAW's new development stewards have been making up for lost time in getting 'timely' updates out; if anything, I wish they'd slow it down just a wee bit; two updates a month would be plenty, but, hey, after its miserable time in the house of Gibson, corporate owners who are committed to maintaining the platform sensibly are a very nice change.)
Old 5 days ago
  #15
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Karloff70's Avatar
 

Quote:
Originally Posted by bgrotto View Post
Absolutely yes. No latency is better than any latency, no matter how small. Excellent musicians can feel the difference between none and “only a little”.
Indeed. It's not even about whether anyone 'can tell'. It's about what it makes them do. And analog foldback gives fullest possible access to connecting and finding into to the pocket.
Old 5 days ago
  #16
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Muser's Avatar
Quote:
Originally Posted by theblue1 View Post
You bet. Having moved to a DAW from a hybrid 16 track MIDI/ADAT tape rig back in 1996, I've seen DAW use go from ultra-tweaky, nobody-breathe precariousness to relatively reasonable dependability and ease of use and even setup. Emphasis on relative.

That said, I nonetheless do a ping loopback test typically once or twice a week when I'm working to make sure things are still on the square. (Updates these days are frequent; occasionally that's meant my settings getting rejiggered. My DAW's new development stewards have been making up for lost time in getting 'timely' updates out; if anything, I wish they'd slow it down just a wee bit; two updates a month would be plenty, but, hey, after its miserable time in the house of Gibson, corporate owners who are committed to maintaining the platform sensibly are a very nice change.)
one of the reasons I have few pieces of gear is so I can know what the factors of that gear are in respect of some manageable degree of certainty regarding its performance. there was a thread recently where some were angry about a Roland boutique of the Juno. they were angry that it also had the same 15ms latency in response to Midi. ooof. maybe it’s inherent in the way the whole system worked as a system. so maybe even though they modeled the unit in analog circuit behavior, they just can't change that behavior. who knows.

I have an external Midi & USB Midi drum unit which has its own sequencer. so I recently embarked on trying to make sure its internal sequencer lined up when it was externally clocked to the DAW. it took days just to figure that one out. I couldn’t work out why it was recording the Midi in the DAW early. I thought there was some weird preemptive Midi recording issue in the DAW. maybe the DAW’s clock generation was early. I eventually figured out that the unit was designed to send its midi notes 1/64th early and it wasn’t the DAW at all. the only way I finally figured that out was, that when I engaged the units drum roll function, that roll Midi signal was not generated early. so the designers had either decided to do that for some reason on the roll, or they just missed the behavior in the design stage.

and you’re right, you never know if some DAW update is going to make something you already tested as working ok,
not work ok any longer. there are too many conditions to test for, so they let the user find the new bug.
Old 5 days ago
  #17
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Quote:
Originally Posted by theblue1 View Post
...Now, for sure 8 ms is not 2 ms (a typical turnaround for post-AD/DA direct monitoring) -- but it's my position that since it's entirely possible for us to monitor with a tiny fraction of that latency (at least using analog wired or using now gray-legality old analog wireless systems), that is the only acceptable baseline. ...
This is going to be an odd thing to question, likely worthy of the trash bin but here goes.
If '2ms is a general through-time -representing approx a 2' delay, a close mic, and playing standing in a room six to ten feet from our amp feels right, the phones mix would actually ahead of the 'live in the room experience.
Old 4 days ago
  #18
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Don't get hung up on the "sound travels X distance in X ms" arguement as our bodies use multiple feedback loops. When they say its in a guitarist's fingers, it really is as there are adjustments of pressure to shape/sustain/mute the note so it's not just about getting the finger in the right place on the fingerboard or plucking the note at the right time. Our nerves in our hands act as a primary feedback loop. Muscle memory is another loop. When we look at printed notes on a page to play and a click or conductor is providing timing cues, there are other parts of our brain at work too. How the musician was trained comes into play as a classical guy will play what is written on the page differently then what a swing band guy will. Same score but different timing results. Precise timing varies with the instrument as with rhythm instruments there is a push pull of exact timing to create a groove and tension and release.

Adding latency means dealing with an additional timing factor and while small latencies have a higher probabilty of exact compensation from the musician, as you increase the latency you also increase the odds of not getting an exact compensation.

Having an analog cue system even with ITB setups may be worth the cost. HDX systems do allow for certain plugs to be used in the cue without a huge latency penalty however with an analog cue it's easy to use outbaord for that function.

It would be nice to have all playing live in a room together like Motown / Fame - Muscle Shoals / Stax / Wrecking Crew - LA / etc. as they seemed to be able to consistantly get a groove going along with push pull timing. Good muscians have worked out this whole timing thing before stepping into the studio. Our job is to not get in the way and an analog cue system is the least intrusive to their multiple timing feedback loops.
Old 4 days ago
  #19
Quote:
Originally Posted by Wayne View Post
This is going to be an odd thing to question, likely worthy of the trash bin but here goes.
If '2ms is a general through-time -representing approx a 2' delay, a close mic, and playing standing in a room six to ten feet from our amp feels right, the phones mix would actually ahead of the 'live in the room experience.
It's not a odd thing to question at all, actually. Your question is entirely sensible given most folks' experience. As I noted earlier, I'm well aware of the sound-in-air math.

There are a couple of things, though; for one thing, I've been overdubbing guitar into headphones since 1981 on a regular basis, typically with the a common close mic placement on the amp (though in the last few years I've begin experimenting more with somewhat more distant miking). For the next decade and change, I used analog decks (which, of course, do not have the scale of latency we associate with AD/DA round trips, even in a dedicated device), headphones fed from the monitor output provided by the simul-sync head. After that I had an ADAT setup and analog board. In '96, I began moving to a 'hybrid' DAW setup, tracking and mixing through analog boards (I had a separate monitor/cue setup).

As noted, I did experiment with monitoring direct out of the my then-new interface when I got a multi-i/o Firewire box around 2004 -- but, what can I tell you, it just felt wrong.

But you know, running a room full of hot gear in when the temperature goes over 100 is really unappetizing -- so I've tried to find ways of working with minimal gear on; but I've yet to be truly comfortable with direct in guitar monitored over cans after AD/DA. (Happily, I do still have a little Mackie 1202 I can press into service in a pinch.)

Vocals and acoustic guitar are much more doable for me because the sound of my own voice or the acoustic occurs when I expect and tends to mask the delayed effect of the source monitoring in such cases.

PS... I was NEVER someone who sat across the room from his guitar amp. It's generally within arm's reach when I'm playing. (In fact, when I saw Buddy Guy in the '80s and he did his guitar walkaround -- still cabled in those days -- finally walking out the front door briefly, I found myself wondering if the acoustic delay ever bothered him.)
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