Help understanding latency please

[RighteousSine]

Hey, recently changed the way I work with my DM3200 and struggling to understand latency and if it's going to be an issue for me and when to use channel delay.

I have 16 inputs from various sound modules and desktop synths, going into line ins 1-16, outputted to Bitwig through FW card, slots 1-16, processed and then returned through channel inserts 1-16 on the DM. Channels 1-16 inverted so I'm listening to processed signal from daw.

Also sending out soft synths via slots 17-32 into channels 17-32 as needed.

Midi mostly coming from an Octatrack

My plan is to mix on the tascam and get a Motu 2408, use tdif to send 24 channels into a separate daw to record as needed. So I want those signals that have passed in and out of bitwig to be in sync when they reach reaper which I'll prob record into as I jam.

Do I need to worry about the latency that's showing on each channel in Bitwig due to vsts or is Bitwig handling that before it sends the audio back to the desk?

Or is that what the channel delay is for and do I need to adjust the channel delays individually in the module/set up page? If so do I just adjust them to the value shown in Bitwig or relative to the track with the lowest latency?

Thanks

 

[mixerizer]

wish i could give more specific advice but i dont use bitwig. from your description, bringing channels out and back in via Firewire will definitely introduce some latency compared to your live jamming, but if the interface ur DAW uses (2408) is on the same computer as ur FW card, perhaps latency compensation / drift correction can be used between the two to reconcile timing. on a mac, thats done by using an aggregate device.

 

[RighteousSine]

Thanks for the reply I'll have a look into it, the more I think about it the more I struggle to understand it and I'm not sure what I'm looking out for, even if I knew what to do about it.

If say I had 2 synths running into the daw and back to the desk, but I was mixing the 8 outputs of a drum machine directly on the desk without going thru the daw, would I need to compensate for latency on the desk then and how would I go about it?
Thanks

 

[mixerizer]

in my experience, its all dependent on your computer and the software u r using to generate the sounds - but if u have a set of synth and instrument inputs from your DAW and a set of local inputs like the drums both coming into the board, the board should be able to handle the sources and any playing and recording in a “live” feeling way. if you do have issues with latency and having them in sync, they will become apparent as soon as u find yourself overdubbing onto the DAW while listening back. That’s a round trip via firewire (latency = DAW playback time delay for what you hear + time delay to put your new recorded material into the DAW).

Using software input monitoring, zero latency monitoring and track lineups/timing analysis techniques, etc., can all help u minimize the timing issues. Note that Firewire, like and moreso than newer protocols, has some timing limitations as to just how fast it can move stuff in, process it, and move it back out - or for overdubbing, process and move DAW audio out to the board and process and move new overdubs back into new DAW tracks as close to immediately as possible.

 

[RighteousSine]

Ideal thanks for the explanation, need to do some more reading, but sounds like fixing any timing issues in the dubbed tracks is easiest unless obviously apparent in the signal chain

 

[mixerizer]

yup, for sure! if u can get your studio to a place where u know where the latency exists and how much of a delay it imposes, u can figure out how best to musically overcome it.

sometimes slipping the new overdubbed tracks a little earlier on the timeline (manual latency compensation) is the right solution as you suggest. sometimes the DAW will accomplish that automatically to some extent. other situations might dictate that you “freeze” effects and tracks on your DAW (which reduces processing delays by printing the plugin and effects to temporary files), to shorten that delay so its close enough that the timing doesnt feel off to your ear.

i try to use “zero latency monitoring” wherever possible - thats when u listen to live audio during recording, to reduce the latency “round trip” to a “one way trip.” in that mode, i listen to the sound of what i am recording in the mixer, directly from the source (not thru the DAW).

U can combine these techniques - and i also push the computer as hard as possible (smallest sample buffer it can manage, etc.) and reduce the other software services and functions running there, which means being on constant guard for pops and clicks from CPU overtaxing.

in the end, though, i have to admit - after starting on TASCAM portastudios, then reel to reels and then after decades of Pro Tools and Logic, I am feeling very enthused about my newest (old meets new school) workflow, using DAWs as sound sources and sequencers, and my DM mixer and x-48 48-track recorder in the recording loop. Recording and overdubbing is essentially zero latency, and I can send tracks into and back out of the various DAWs for deeper editing or dsp, and its easier to hear the delay each varying plugin or softsynth operates at.

Latency has been an issue forever - its one of the reasons i feel like TASCAM hardware digital mixing is/was a sweet spot of audio technology that hasn't been achieved since at this price point - it’s pro audio, its’s a physical system with reliability and software independence, and having come up in music recording with professional studios as my “role models” for my own setup, hardware recording and mixing of software and hardware instruments feels more organic, immediate and robust than using a DAW for all of it (instruments, mixer and the recorder).