Perhaps utilizing the WesAudio ngLEVELER at the end of your individual insert chains, you could get automation post FX but pre-fader on your API console. That's what I would do if I had enough inserts on my analog summer. Thanks for your informative videos, much appreciated!!
Why not set up group tracks in the DAW....automate your tracks going to the groups/stems. Then send the stems to the API including the drums stem. Use the x-desk for effects returns and alike and stem that to the API. First 8 channels on the APi would require inserts so rack the 500 modules. 24 channels with inserts on the mix with stems. Automate the groups as required. You have lots of choice gear to insert as you desire. Just an idea that maximises the API flavour on all the mix. Now the next trick is to record also and you have it all. Love your posts they inspire. Cheers
Hey, thanks for sharing. I don't really understand what you're describing. I'd be sending the SSL to eight more channels on the API? The system that I have now, I'm using API EQ on the SSL channels and then the SSL is feeding the mix buss on the API so it's hitting the only transformers there are on the API. I could use the SSL for effect returns but honestly it's more useful for Drums because it has its own bus insert. This way I have two stereo FX sends, individual channel inserts, bus insert and 2 stereo fx returns all dedicated to the drums. It would be kind of a waste to have that on fx returns. If/when I ever step up to a 1608, then I could see using the SSL for effect returns. The other thing is with my current set up I have 32 channels that all have inserts, And I find myself using them a lot lol.
Sum the SSL to a two channel stem to the API on two channels only. As I understand the API first 8 channels cannot have inserts in the first 8 if your eq's are installed there? 24 channels all dedicated to stems on summing is more than plenty. A drum mix stem on the API returned to the DAW via the SSL in stem or track for track would yeild a sonic flavour. I find that SSL at the end adds so much goodness. Pushing the faders after valve & transformer , eq and compression etc you think you are there but then there is the Super Analogue sprinkles. Nice. Recording or grouping stems reduces the recall it's in the DAW so it's recalled. Out board gear think certain equipment on certain stems the patch bay can be preset and changed as required for something other than the pre configured/prefered as required. Minimum recall. Just sharing ideas...I went small desk Calrec for summing stems/recording and creating stems and SSL 6 as the utility parallel compression Stem recording into an apollo 16 . Valve mic pres Analoge addicts Joe Meek Twin Q2 Neve VR Labo K 2 Mic pre eq and dynamics Amek 9098 mic pre + 2 pres on the SSL 6 and 8 Mic pres on the Calre . I get 24 summing channels combined SSL/ Calrec. I only use 12 for stems on the Calrec I found the SSL adds more at the end of the line driving the fader after valve transformers and then there is more. The SSL 6 is very pro kit. Your x desk has that Super Analogue goodness too. Regards
Ah, ok that's not correct. you can have inserts on the first 8 even with the 500 series EQs installed. I am summing The SSL to a stereo input on the API. it's a special injection point on the master bus that is before the master bus insert (and the built-in compressor). It's literally called "Summing Input". it's a fantastic feature that they actually didn't put in the large consoles, which is interesting lol.
Seems you can have 32 outs to your interface using inserts on the summing 16 side of the box + what you have on the x desk direct out plus the left hand side of the box. Normaled insert send to your interface on the summing side of the box gets you there. The return through dedicated hardware or a patch arrangement get you transformer valve eq compression on mix down per channel. If you create groups/submixes/bus's in your DAW you can leave your self only recall on your hardware. The Aux set up across the DAW I/Os is well worth exploring. Cheers.
then you need double the channels of interface. 64. and it's also getting the audio converted twice. It's too convoluted of a system. The real answer just to have automation on analog faders and use it the way it's designed to be used. in the meantime, I can auto punch my mixes as I'm capturing them, and change the analog faders manually. It's faster, easier and sounds better. even in the best case scenario, people that are using aux sends, whether it's in the DAW, or the analog inserts are not getting optimized conversion levels. you just do it on the mixer manually, you have one well optimized digital to Audio conversion, and analog processing is optimized... its just better
I don’t know if we watched he same video or not, but after dreading recalls but refusing to stop using a console for a decade now, this method has absolutely changed the game for me. I had to align the console with the two ad/da boxes, but so far it’s been great. I don’t do daw inserts, though, I just mix, then bounce the tracks back to the daw. Makes me wonder why people much smarter than I have long since gotten rid of their consoles because of recalls. It sounds like you could potentially record the console outs AS you’re doing the automation and 2 buss mix, no?
Sure. it saves another pass. I would really love to get the DAW automation after the analog processing, but I don't see any way to do that without doubling the channel count. On pop mixes I need all my channels.
@@hopetownsound ah, I see what you mean now. One theory I have with having daw automation post-console is to double the DAW tracks, monitor the inputs, write the automation, and return the daw 2 buss output to a tape return on the console. I’ve never tried this though. I guess this would hinge on being able to monitor daw inputs while it’s reading automation.
right, this is why you need double the inputs. And you're summing the monitor inputs > master channel > DAW in / Mix capture > console 2 Trk input for mix monitor. So 32 in/out converter + additional stereo A/D/A converter. 64 analog channel inputs, 34 converter channels... its a lot.
@@hopetownsound The only alternative I can think of would look like with 32 I/O: Channels 1-30 digital out > console channels in > channels 1-30 digital in > daw channels 1-30 with Record Monitor on > digitally summed to daw 2 buss > channels 31-32 digital out > 2 track mix monitor console input. So I guess the trade off would be that you’d be recording a digital sum and I’m not sure if the I/o delay would be an issue.
A common variation of this is: 32x32 Interface: DA -> 32-Console Channels -> AD. Either as Hardware Inserts on 1 DAW Channels 1-32, or Split across 2 DAW Outs/Ins Console Sends on 1-32, Console Returns on 33-64 (MTK Tape Machine setup.) Then… 2x4 Interface: DA -> Console 2-Bus (+ Inserts of course) -> AD (2-Track Print) The benefits of using separate Interfaces are: Using different 2-Channel interfaces that are relatively inexpensive and can have very different sounds. Also, you can actually Record your mix to a different Computer altogether, like a Mastering Setup, allowing you to also change the Bit Depth and Sample Rate of the Master Print. Say your tracks are 24/44.1, after all of your analog processing, you may wanna capture at 32-bit/ 192Khz. You may also want to print multiple times at different resolutions. And you may also wanna Split your Main Outs into multiple 2-Channel Interfaces, say an SSL 2+ with and without 4K, Audient, AMS Neve 88 with Marinair Transformers, a pristine RME or Lynx or Grace or Lavry, whatcha got.
cool! That was a great introduction. I'm a live sound system engineer. I've been studying the origins and mechanisms of these systems lately. Am I correct in thinking that before there were mixing consoles, mixing desks, there were individual preamps for each source, and then there was a line mixer (without preamps) that brought those sources together, and then the line mixer had the role of suppressing and mixing? In other words, there wasn't a console-type mixer right from the start.
I have to admit, I don't know the history of that as well as I would like to. I know Bill Putnam was one of the early innovators building consoles and obviously eventually Abbey Road. i've done my own recording that way since 2000 .
Two cents: Abbey Road in England was a good decade and a half before Universal Recording in Chicago, and leaving Rupert out of the discussion is probably audio blasphemy, however Bill Putnam is credited with among Many things, laying out Recording Consoles as we know them. This vide touches on, without directly addressing, the difference between Split and Inline Consoles. Along the lines of your question, this is an oversimplification but, you can think of a Split Console as two separate boards. The first on the left, has Mic & Line inputs to each channel and feeds each channel individually to a Multitrack tape machine’s Record head. The other on the right, has inputs coming from the Multitrack’s Playback head, and mixes Dow to the 2-Track Input Path, Monitor Path. The Artist Cue mixes happen on the Input Side to avoid the Tape Delay (OG Latency). The Producer/Engineer needs to hear what’s being Recorded, so the Monitor path. Inline Consoles (credited to Harrison) put BOTH paths in One Channel. A dual-path, dual-input Channel Strip. The API Box-2 is really an 8-channel Split Console on the left (with a great set of I/O on those 8 inputs, but still only one path), and a 16-channel Monitor Mixer (aka Summing Mixer) on the right (again with a comprehensive set of controls for a monitor mixer). The SSL X-Desk on the other hand is an 8-channel Inline Console (again with great I/O and Routing on DB25s). Basically the X-Desk has the functionality of the Right half of the Box 2 during mixing, and 8-channels of Inline functionality while tracking, minus Mic Pres. But not both at the same time. As far as I/O, it would not be difficult with what is in this video already to setup these two desks plus patchbays in a Split Console configuration, where you print everything through to tracks on Hardware Inserts, and then either sum ITB and Hardware Insert out through API 2-Bus processing and back in to print. Or spill Everything back out for Analog Summing to the 2-Bus Chain, back in. Assuming you have enough conversion. As far as the Sound….that is obviously subjective, and based on preferences. But assuming that the Analog pathways of the DAW, the Superanalogue SSL, and the 21st Century API architecture, minus any additional processing, are all so neutral that you could Null Test em down to -100dB, Then the Sound your going to get comes from specific Stages of your signal chains, and what you choose to mix through. The SSL has No processing on-board and by design is Absolutely colorless. The Box has 8- API 312 Mic Preamps, and 2- API 500-series Compressors as a Stereo Bus Comp. I don’t believe there are Any transformers, but the 2-bus is running through API OpAmps? Correct me if I’m wrong. The Box was basically a package deal that got you most of what you need for a decent price, so that you didn’t have to piece it together yourself. However, the problem arises when you outgrow the Box (pun intended). You can definitely build around it, which you are doing, but I’ve worked in a studio with an SSL X-Desk plus X-Panda, that had racks of gear for the front and back end. That includes API Pres and API 2500s, etc. (Plus a whole lot of CAPI) and you may disagree, but to my ears, That will get you the API sound. I’ll revisit this with more time, I have a couple of other thoughts down that branch of the tree, but to the original commenter’s point…. Yes, we went from a bunch of pieces -> to comprehensive Consoles -> back to pieces and signal chains. Some of us Always thought of the board as a bunch of parts anyway, so this isn’t an entirely new phenomenon. Running Patchbays for every conceivable in and out is the elegant way of making everything in the studio Modular. Adding 1 or MORE computers to the equation only begs the questions: “how much conversion do I have?” And “how should I Normal everything, In and Out of the box(s)”
well, there's no such thing as a floating point converter, you just have to be careful and watch your levels the same way you would anytime you're recording.
