Latency

[dans]

For the last three months to see if I liked PT software, I used a "borrowed" copy of PT 4.3 with my BW 350, 128MB RAM, second video card, Orange 906F SCSI card connected to a Seagate Baracudda, and only the internal 1/8" sound jacks. I have no recollection of EVER hearing a noticeable delay between my playing and hearing a sound.

I bought the 001 and added another 128MB of RAM (256 total, huge amounts allocated to PT and DAE, can one allocate *too much* memory?). I immediately noticed a very long delay when recording. I reduced the hardware buffer and it is better (CPU usage is set to 75%). At the lowest setting I don't think I perceive a delay. All of the other settings I find unacceptable. It begs the following questions:

1) How many tracks will I be able to use with the buffer set to 128?
2) Why did my 4.3 software and the built-in hardware perform so well vs. the 001?
3) Am I too sensitive to this latency and doomed to never being able to use a host-based system?
4) Does the very small delay of 2.9ms at 128 samples affect how one plays even if s/he doesn't THINK it is noticeable?
5) What are some drawbacks to using an outboard mixer for monitoring, as suggested in the manual? If a mixer solves the problem, then I'm off to Guitar Center ASAP. Any mixer recommendations? Mackie 1604? Are 4 busses enough?
6) What other pieces of gear should I get to complete this system?

I apologise if these are elementary questions. I am new to DAW, but not to music!

[This message has been edited by dans (edited December 13, 1999).]

[This message has been edited by dans (edited December 13, 1999).]

[JMan]

The latency seems to be the biggest issue concerning the 001. First of all, PowerMix needs only 48 megs and LE needs 128 megs (192 recommended). So, the software is a considerable amount more indepth. There are 8 more tracks, RTAS and AS plugs, MIDI, and more I/O options.

I personally don't care for Mackie mixers. This might iritate some people out there, but I am entitled to my opinion. I like the Alesis Studio 32. Direct outs on every channel, 4 buss, and about $150 cheaper than a Mackie. The Studio 32 is about $750.00. Allen & Heath make good mixers as well.

[Chris Coleman]

as far as input monitor mixing goes, I use a little Behringer MX 802A. 4 mic/line channels, 2 stereo channels, great qualilty. I've also got a Mackie 1604VLZ, but that is usually hooked up to my DA-38 (which is why I bought the small Behringer).

[dans]

I appreciate the responses. But what I really want to know is what are the downsides to using an outboard mixer? It seems like everything in PT has advantages and disadvantages. Lower the buffer and you get lower latency but fewer plug-ins/tracks; raise the buffer and you get more plug-in capability and tracks but unacceptable latency.

So again I ask, what are the trade-offs in getting an outboard mixer to reduce latency? Will I actually be playing a few ms ahead of the rest of the recording???

I am willing to spend more cash to get this thing working well. I'm just not sure where to spend it.

Thanks for any and all help.

[Darrell Diaz]

Dans,



The drwabacks to using an external mixer are 1) A more complicated signal path, 2) possibly more noise 3) more money spent. I've tried using my 2408 with no mixer, but the latency even at the lower buffer setting made my MIDI track late by a small amount and I had to shift all MIDI tracks after recording just to maintain the feel. It was too much extra work. (The was even a noticable delay using the SPDIF out on my Proteus 2000.) My current solution is using a Mackie LM-3204 for all of my synth inputs, DAT, cassette, etc. and busing into my AMIII card via the Alt 3/4 bus. This solution allows me to hear all of my inputs even with the computer off and also to EQ, effect and adjust the gain of all my synths, mics (2 extra mic pres on the Mackie), guitars, etc. before they hit the A/D converters. The other option for you is to record with the lowest possible buffer size, then adjust the buffer setting higher for more FX while mixing. Hope this helps.

[Stenjy]

I've been loading previously-recorded ADAT sessions and mixing in PT 5.0LE since I got the 001. Yesterday I finally get my recording setup all patched together and lo and behold, I found out the definition of latency. My question is, how do you hook up an external mixer to solve the latency problem? I just wanna be able to give clents a true read through headphones and not feel like I've got the magic "slapback" button engaged. I've got a couple of Mackie 1604's. Is the latency problem on the inbound end, or the outbound end? What I was hearing is almost unusable. I understand setting buffers, and will try that, but if I can solve the problem with a 1604, I can handle the latency in overdubs. thanks and Merry Christmas.

[lwilliam]

I haven't found the latency an issue with the 001 (128 samples buffer), but when I used my 1212 card with Cubase, it was unusable.

All I did was to monitor everything through my Mackie 32x8 just as I did prior to "going digital". Usually, I was recording one or two mics at a time and I just split the output of my outboard mic pre (or D/I) with a Y cord: 1. to the 1212 input(s);
2. to a channel on the board.
A group of mult jacks on your patchbay will work just as well - possibly even easier.

For playback, I had the 1212 outputs sent to a channel pair on the Mackie board and muted or unmuted the channels as necessary. The Y cord is a little hoaky, but it worked. You've got to remember NOT to monitor the input to the 001 or you'll get a feedback loop going.

The main disadvantage of using a mixer for monitoring (IMHO) is that you'll still have to keep, and set, your outboard processors (reverb, etc.) so the headphone mix sounds reasonable to the performer(s).




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Larry W.

[CCash]

I hope someone can answer the latency question regarding how and where it's happening. I don't have any problems now with the Toolbox (AMIII card). Or maybe I never noticed it. Did anyone making the AMIII switch to the 001 notice a new latency problem?

I have two choices for recording path; to a DA-38 or to ProTools via AMIII, or both (they're locked). When I record I want myself and the performer to know what's really going down to "tape", so I monitor the computer or DA-38 outputs, almost always. Either machine is record enabled, and of course the signal passes through to the mixer.

I want to know what new issues would crop up for me with the 001 or PTle. If you record enable a track in PT, and monitor off of the output you send it to (bus it to), are you saying you hear latency? e.g. You hit your guitar strings and what you hear in the phones is lagging. That's not good. With the Y-cord/patchbay fix, or just monitoring before it hits the computer, is the problem solved or does the track still go down late? i.e., is the latency only in monitoring through the computer??

If the tracks go down spot-on, then monitoring your take pre-computer solves that. However, if the take is actually recorded late, then you'd still have to move the track forward after recording (assuming you can hear the delay).

The analog equivalent to this question is, Is the delay happening at the record head or the play head? It's the $3,000 question.




[This message has been edited by CCash (edited 12-28-99).]

[dans]

CCash: precisely my question!

I never specifically asked this, but that is in essence what I want to know. Is the delay happening at the play head (bad news), or the record head?

Thanks for putting the correct words in my mouth.

lwilliam: could you explain the "Y" scenario a bit more? Why can't I just get an outboard mixer, monitor through that, and have the 1st 8 channels go direct into the 001?

thanks for the help

[lwilliam]

Dans:

You can certainly get an outboard mixer and monitor everything through that (using the direct outs or busses to feed the Digi001). I generally try to minimize the signal path when recording by using a high quality external mic pre and/or Direct box without going through the electronics in the board. That's the reason for the mult or "Y" cord.

You do have to consider when you are overdubbing: how will you monitor the existing digital audio in addition to the "new" signal going to the 001. As long as your "new" signal isn't being passed through the 001 back the board (where you would be listening to both the "new" signal and the "latent" signal simultaneously), then you're fine.

As far as the "BIG" question posed above, there are some in this forum who are much better versed in the electronics side of things than myself, but the basics are that ANY A/D conversion will have an inherent latency and ANY D/A conversion will also have an inherent latency. I understand that this latency is in the 1-2ms range and would be experienced with ANY device that converts analog to digital or vice versa (any DSP device, digital keyboard, sampler, etc.). However, in addition to the inherent latency in the conversion process, there would be an additional latency caused by the host processor and any inserts to the signal chain. This is where it can really become noticeable. That latency can make the signal virtually unusable for monitoring if it reaches the 20-40ms range. The signal would sound either flanged or like a slapback delay - if you listened to it with the original (non-delayed) signal. However, the "round-trip" latency of 3.9ms for the 001 seems rather negligible to me - others may disagree. Remember that ALL tracks are going to have the inherent D/A delay of 1-2ms if you are listening to them. That effectively cuts the 3.9ms round-trip delay in approximately half.

Hopefully this clears things up a bit. However, if any of the electrical engineering types who read this forum can be more precise than my figures above, please chime in. I got my numbers from a few articles I've read here and there and don't pretend to be an expert in that area.




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Larry W.