Mixing in Scope - Device latencies/phase problems

[Yogimeister]

And here is another take on it (until I load my pics and audio which I will arrange tomorrow)


Lets say you are recording a drum kit (i font do this, i use samples - so this is in theory for me)

You have 4 mics ... Bassdrum, snare, toms and overhead ... (Of course there is a bit of all the drums in all the mics ....)


You track the mics with no fx into cubase ... <align any phase offsets cause by the different mics/preamps/signalpaths> ... Then send the channels to scope for a mixdown ...

Now lets say you just want to compress the overhead/room mic with two compressors and then shape it with an EQ. And you put another compressor on the snare .... (All inside SCOPE)


Wouldnt this setup cause phasing between the different channels ? (eg, the snare and hihat captured by the overheads are out of phase with the snare and hats from the snare mic - and both are out of phase with the tom mic ... )



How would you go about doing this ?

[garyb]

align phase on the drumset mics???!!!

[astroman]

I'll quote myself ...

you really should not take it too serious - it can drive you nuts to align everything down to the sample level
it won't sound better because it is aligned (in fact many sounds need some 'displacement' to please the ear)

cheers, Tom

[hubird]

For your information, I never noticed anything wrong regarding phase problems, not with band mixes nor with electronic music, when using my setup with a DAW mac and a cards mac connected through adat.

I put plug-ins everywhere, channel, master, or cabled inside the project window, it doesn't matter, and I have good monitoring.
It should just work
I just make 100% corrections for latency delays of re-recorded tracks.

You could try running the project on one card, if you have more? If not Xite...
Smart sound examples (single track/total mix) would also be of help

[Mr Arkadin]

Like hubird, and many others I suppose, I never even think about it. All sorts of routing going on, bussing out of Cubase etc. and I have never noticed it (maybe my mixes are shit?) and would certainly not consider analysing it to the extent you are suggesting.

Maybe an example of you project and some sound examples might enlighten.

[Yogimeister]

Before I head to bed - here is an example of my routing from earlier in the project
(When I noticed that bypassing the LA on the kick (when it wasnt even compressing much) affected the highs in rest of the mix for some reason ....).

(I took a snapshot, disabled the compressor and moved on ... But then realized the problem is probably phasing .... )


I will post sound examples tomorrow ...



In the pic you can see the busses (and nobus) going into the "second summing mixer" (the asio ins are just for resampling ).

Master chain is on top

[hubird]

smaller...

[Mr Arkadin]

Any chance of a re-size?

Also, I don't think it would make any difference in Scope, but why not use the ASIO2 modules trather than the ASIO1?

[garyb]

yeah, well, i think if there's really a good reason to work like that, then dealing with possible phase issues just goes with the territory.

just because you can, doesn't always make it a good idea. i think there might be serious phase considerations if these techniques were used in a hardware studio.

[garyb]

you need a correlation meter/oscilloscope...

[Yogimeister]

Sorry for the size - its straight from my iPhone ... I will resize next time.

I dont use ASIO2 because I dont need the extra features listed in the user manual (I think ...) - which is monitoring and stuff like that .... Would it make a difference with my current setup (I dont have any HW fx ... And I monitor my mic recording via Scope when I record vocals)

Im not really sure what is so "different/extreme" about my layout ... (All I wanted to do was group channels together so I can apply common effects like SC and bus compression / bus EQ .... Is that freaky ???


(I think I would be having the same problems if I didnt use the busses at all and just added fx to individual channels .... No ?

Sound example will be coming later today ...

[Yogimeister]

I put plug-ins everywhere, channel, master, or cabled inside the project window, it doesn't matter, and I have good monitoring.
It should just work
I just make 100% corrections for latency delays of re-recorded tracks.

What do you mean by 100% corrections for latency ?

Do you perform the "mix" with an empty mixer after you re-record the effected tracks to your DAW and correct their offset ?

Do you correct each channel for the latency caused by the round trip latency AND the scope devices placed on that channel?

[hubird]

I just align the audio of a re-recorded track with the original one.
Nothing serious, just some samples to the left

[astroman]

that's the method I use to check if bleeding between microphones is disturbing or not
(pick a prominent peek of the waveform, zoom in and shift)
if sound improves, I leave it - otherwise revert to the previous state (but only on 'important' signals)

the 'problem' here is a highly specific case regarding processing of layered drum sounds
(looks like I misunderstood the original question)

Scope's bag of tricks on regular drums to yield a fake 'Steven Slate' result... is exactly as you describe it
it's 1 sample here and another there in the DSP routing (you can chain/parallel arbitrarily...)
and indeed: Scope projects don't always keep the state you tweaked out of it
the project may change in this regard - not necessarily, but it can happen

imho it's the same in native processing, probably even more intense - but nobody can look inside...
(as you can with Scope)
Steven Slate uses dedicated processing within 1 device to align multiple samples for drum layers in Trigger
(the software aligns transients, not regions)
for sampled sounds I'd rather switch to such a device than to do it by hand

cheers, Tom

[djmicron]

beware of stereo kicks, i think you should take care to the mid/side to avoid phase issues, it's not all related to the processing latency.

[dawman]

Stereo Kicks are lame, they make a real drum set sound like Kontakt.

I actually use Brainrox BX Digital 2.0 Mono maker for all freq's below 220k.
Worked so good recording I tried it live and on a pair of QSC-K12s it makes my synths and sample punchier will a better low end focus.
Don't need it on a CPS monitor though.

[garyb]

lots of stereo samples is just not smart as well. how can you create a nice stereo field with a reverb if there are 20 different stereo fields competing within it? there are bound to be phase issues that way.

[Yogimeister]

Yeah, I usually mono my kick below around 150 and my bass below 60 - but I guess I might be abusing stereo a bit ... I just like to have some sounds centered - with a bit of stereo - and some sounds spread out more - kindof wall-of-sound-ish ...




I wasnt able to reproduce a noticable phasing effect today with my project (not sure which files recreate it though - so it might be a case of the DSP positionings of the fx ....)



One thing I guess I could do is assign the mixer to specific boards (I currently dont) ... Will that help ?
(I have a luna and 2 pulsars)

Also,
Regarding the ASIO IOs - I have read the manual & searched through the forum but still have a few questions ...

I am running cubase 5.51 (at 44.1K) , recording in cubase at 24bit - with "ASIO1 32" (As recommended in the manual, though cubase is floating point AFAIK ....)

1) Should I use ASIO flt ? (Will it make a serious sonic difference ? Can I still record in 24Bit or should I dither down to 24 ?
(Or is it advisable to use ASIOflt and record at 32flt ?)
2) What is the internal bit depth of scope processing ?
3) should I use ASIO2 rather than ASIO1 ? (I dont do direct monitoring, use ADATs or use multiple ASIO apps) - will it effect the sync and sync stability (with cubase) ? What should I connect to the clk input on the ASIO2 dest (or is it an ADAT thing?)



Sorry for the drilling - and thank you so much for your help ...
Have a wonderful new year !

[astroman]

Scope's standard format is 32bit integer for internal communication
some devices extend precision (for temporary calculations) to a 40bit float format (32 bit + 8bit exponent)
internally it's best captured wth VDAT that will record the original DSP data stream (afaik)

at least I can tell that tracking with VDAT plays back absolutely identical (in case Scope feeds the monitors)
any other software, be it Wave drivers or Asio that run through native processing alters the sound
(people say that from the math side it's impossible... but that's what ears will tell you)
the resolution of the driver doesn't matter in this context - 16bit will do it as will 32bit float
(the latter is a 24bit integer + 8 bit exponent number)

unless you're recording orchestra you can safely ignore dither - it won't change quality
on anything from electro, funk, rock, pop, metal it's just a waste of effort
other parts of the processing have way more impact on the result (imho)

get some idea about the dynamic range from 24 to 32bit in relation to regular 16 bit playback
(which is still standard and covers a range of roughly 90db)
if 1V represents the full signal, than the 24th bit represents something like 0,001 mV or so
(that about dithering from 32 to 24)
I mostly use a 24bit internal project format and just write the stuff as 16bit files for output
(for my ears there's not even a faint difference - maybe I'm already deaf...)

Scope is frequently experienced to sound better, more 'clear' with better impact etc
imo a native sound engine will kind of drift with CPU load/interupts/tasks or whatever
(those chips are incredibly complex - and I'd never trust any measurement of them)

bottom line: use the driver that's the most convient for you

cheers, Tom

[dante]

lots of stereo samples is just not smart as well. how can you create a nice stereo field with a reverb if there are 20 different stereo fields competing within it? there are bound to be phase issues that way.

I agree. Why don't you try the CW16x16 which has 16 mono and 16 stereo.

In fact my mixer has 32 mono channels so I always start with the mono stuff first, and maybe end up with 8 mono then pair up the remainder to get up to 12 x stereo.

And those 12 stereo are not all panned hard left and right.