lapslap recording sessions, september 2010

lapslap are Michael Edwards (saxophones, laptop), Martin Parker (horns, laptop), and Karin Schistek (piano, Nord synthesiser). We recorded free improvisations in the Reid Hall from the 8th to the 12th of September 2010. This was to be our fourth album on Leo Records. On September 11th, the American percussionist and 9/11 survivor Fritz Welch visited us and guested on a 3-hour recording session.

Aware in advance that the decision was insane, Martin and I decided to engineer the recordings ourselves, despite being performers. We wanted to put the new studio through its paces and try out some of its more esoteric possibilities, e.g. ADAT sends from the studio to the hall via the CAT 5 extenders. We couldn’t inflict that and the up to 12-hour days on some poor unsuspecting engineer….

setup

microphones

In contrast to our last recording–where we aimed at maximum separation of the instruments to isolate signals–we took the counter-intuitive approach of placing the instruments close together, relying on spot mics for separation where necessary. The theory was that, whatever bleed happened, say, from the sax into the piano mics, if they were close then we wouldn’t have strange-sounding delays i.e. the bleeding signal (if you will) would at least be direct, not reverberant. Having listened to the results I have to say this worked.

We wanted to capture the acoustic instruments with both close and distant miking techniques. To this end we used Neumann U89s in cardioid mode as the main air mics on the sax and horn. These were placed about 15-30 centimetres from the horns’ bells and they were isolated with SE Reflexion Filters. As we’re fans of double-miking horns etc. Martin also used his beloved DPA 4061 omni as a clip-on whereas I opted for an Electrovoice RE-20 as my second on the saxes.

The piano was close-miked with two AKG 414s (in wide cardioid mode) about 15 centimetres from the bass and treble strings. These were mainly aimed at picking up Karin’s inside-piano effects but also blend nicely with other mics to vary the closeness of the sound (if you’re careful to avoid or deal with the proximity effect). We used two Schoeps omnis on the piano as well. These were spaced just over a metre apart and a similar distance away from the piano. Clearly these picked up a considerable amount of horn signals and room as well, but the omni pattern at this distance transduced some amazing bass from the Steinway piano.

The ensemble was captured by a central Schoeps mid-side coincident pair with the cardioid focussed on the piano. The pair was about two metres high and perhaps three metres away from the piano and horns. The horns had clear sight lines to the mic pair and were facing pretty much straight into the sides of the figure-of-eight.

The last pair of Schoeps omnis was in row three of the audience seating, about five seats or so in. The main aim of these was to pick up hall ambience and if we’re lucky these will allow us to get away with adding no artificial reverb to the instrument signals. (It should be added, however, that we did put Altiverb reverb on the laptop signals when we monitored during playing. This really helped the performances and without it the laptop processing would have been too dry to interact with sensitively. We recorded these signals dry though, and will add varying degrees of reverb to them in the mix, perhaps even using Altiverb again with an impulse response of the hall which we took at the close of the final session.)

Placing Fritz was difficult as there was no room in the centre for him to set up the percussion. We opted to put him on the rear raised stage, about 3 meters behind the piano, with clear sight lines to the mid-side and ambient mics. We used a Neumann U89 as an overhead (Martin donated his and took the Neumann TLM 103 I’d brought in for his horn) along with a Sennheiser MD421 Mk2. I was surprised how well matched these were as a pair, actually.

aggregate device

Despite the studio being designed as a 16 channel system we actually recorded 24 channels. This would not be possible on our ProTools HD system (16 channels total with the two Myteks with ProTools cards), but as the 8-channel limitations of the ProTools Core Audio driver had forced us to buy a Lynx AES-16e-SRC card anyway (to support Logic, Nuendo etc.) we opted to record on Nuendo (my preferred DAW) using an Aggregate Device made up of the Lynx and the TC Konnekt 32. We’d mainly thought of the latter as a digital format converter but it’s actually a pretty solid 16-channel firewire sound card too.

I have to admit that I wasn’t absolutely confident that this approach would work so we had a backup plan involving a separate computer to record the 8-channel ADAT stream from the laptops. The idea was to sync the two recording systems through such a primitive device as a hand clap, aligning these when merging the tracks onto one system. Thankfully we didn’t have to do this.

The Aggregate Device (now available to all you Logic and Nuendo users as the “LynxTC” device on the studio Mac) was rock solid. It offers 32 channels of digital input and output over Firewire and AES via the patch bays. The first 16 channels are the Lynx card, the second 16 are the Konnekt 32.

routing

Below are my pre-production notes. Martin and I both have the possibility in Max/MSP to output 4 channels for quadraphonic playback. Thinking of a possible future surround release we decided to capture these rather than record a stereo mixdown.

 

24 input channels needed: 4 piano, 2 synthesiser, 2 horn, 2
sax, 2 percussion, 2 room, 2 distant ambient, 4 martin
laptop, 4 michael laptop (16 mic inputs, 8 digital over the
ADAT extenders)

so we'll need an aggregate device made from the lynx and the
tc konnekt to make 24 channels total I/O.

first 8 mics go straight into desk, from there to mytek 1
and from there to lynx at 96k (clocked from mytek1)

last 8 mics go into desk, from there to mytek 2 and from
there into tc konnekt (running at 96k also, clocked from
mytek1)

8 digital go from martin's fface to adat extenders, from
there to the adat->aes converter and from there into the
lynx, using SRC on the lynx for 48k to 96k conversion.

compression: we'll put limiters on one sax and horn mic (SSL
dynamics 1&2), and on the piano close mics (Vertigo).
threshold c. -3DBFS, ratio c. 10:1, fastest attack poss, so
it's only there as a protection should we get an unexpected
transient.

we use the desk's track busses 1-4 for each of the
instruments' mono mixes (for laptop processing and
headphones monitoring), routing these to the 3rd mytek
running at 48k (internal clock).

nuendo is set up to do the headphones mix of the max signals
over the desk's track busses 5&6, also routing to the 3rd
mytek.

3rd mytek then sends 6 AES to the adat-aes convertor and
into the extender back up to the hall.

so the adat loop is: michael 4 channels of maxmsp to martin,
martin adds his 4 channels and sends all 8 over adat
extender.  studio routes back over the adat extender 4 mono
channels of instruments for processing plus a stereo
headphone mix of the laptops.  this goes to michael's adat
in and he routes the 4 monos to martin along with the 4
laptop; the headphones mix goes out michael's analogue outs
to the headphone amp:

IN from extender
michael 1: piano
michael 2: horn
michael 3: sax
michael 4: guest
michael 5: headphones L -> analogue out
michael 6: headphones R -> analogue out
michael 7
michael 8

OUT to martin
michael 1: piano
michael 2: horn
michael 3: sax
michael 4: guest
michael 5: max 1
michael 6: max 2
michael 7: max 3
michael 8: max 4

IN from michael
martin 1: piano
martin 2: horn
martin 3: sax
martin 4: guest
martin 5: michael max 1
martin 6: michael max 2
martin 7: michael max 3
martin 8: michael max 4

OUT to extender
martin 1: martin max 1
martin 2: martin max 2
martin 3: martin max 3
martin 4: martin max 4
martin 5: michael max 1
martin 6: michael max 2
martin 7: michael max 3
martin 8: michael max 4


clocking

Obviously, running essentially two digital systems at two different sampling rates is not ideal. We had to do this though as we wanted the sonic benefit of recording the mic signals at 96k, even though the laptops were limited to 48k (any higher and the CPU couldn’t cope with what we needed to do). However, the 96k system (Lynx, Konnekt 32, two Mytek convertors) is all clocked from the first Mytek. The 48k ADAT system (Mytek 3, ADAT->AES convertor, two laptops) is clocked from the third Mytek and feeds into the Lynx, which does sampling-rate conversion (SRC) from 48k up to the recorded 96k. This is the reason we couldn’t route the ADAT signal into the Konnekt 32. This would be the ideal choice if everything was at the same sampling rate, because the Konnekt 32 would do the ADAT to AES conversion for us. As it has SRC on its inputs too, we thought we could use it even with the two sampling rates, but it turned out that as soon as we ran it at 96k, it thought ADAT signals needed to be S/MUX’ed so our signals got munged. The Lynx with SRC turned on was the way to go then.

Beware though: the Lynx only does SRC on inputs, not outputs. I was hoping it would be bidirectional so, for instance, in a 96k session we could still use, say, a Fireworx FX processor running at its maximum 48k i.e. SRC’ing both out and in. Not possible I’m afraid.

You might still have expected–and I did wonder–that coupling the separately clocked 48k and 96k systems via the Lynx–even with SRC–might cause dropouts and other nasty little clocking problems. But it seems the Lynx handles this perfectly and whatever it does with the incoming clock and the external clock that’s driving it, it works. Once the system was up and running it didn’t give us a single problem.

clocking tips

When the TC Konnekt 32 firewire cable is in (i.e. when it’s running as a sound card and not just a digital format converter) you can’t change the clock source and sampling rate settings on the front of the hardware. Instead, you have to change them in software, with the TC Near Control Panel (in the Applications folder), on the System Settings page. Set it (and Mytek 2 if you’re using it) to the sampling rate of Mytek 1, and the clock source to external word clock. Similarly, change the Lynx clocking to external and set its sampling rate in the Lynx control panel. (Both this and the TC Near software will start up automatically on the studio Mac.)

If you change the sampling rate of e.g. Mytek 1, you have to change it in the TC Near and Lynx control panels too (and Mytek 2 if appropriate). Always make sure all systems are running at the same sampling rate (unless you’re using SRC). If you don’t, you may not immediately notice problems, but you’ll probably find drop-outs (perhaps as long as half a second), digital burbles or pops, or various other nasty things creeping into what should be a pristine recording.

If you’re having problems getting the TC Konnekt 32 to work as a sound card and locking properly in the LynxTC aggregate 32-channel device, open Nuendo or Logic and first load the TC Near driver as if you were only going to use it as a 16-channel firewire sound card. The sampling rate should then be alignable with the Lynx and Myteks. If that works you can load the LynxTC Aggregate Device and all 32 channels should appear.

conclusion

The studio is now fully 24-channel compatible and sounds fantastic. Really, I’m not sure I’ve ever heard anything sound better than this. The combination of top-notch mics, SSL pre-amps and analogue processing, Mytek AD/DA conversion, and the PMC speakers, is a real winner.

We’re going to edit the sessions in our home studios using Nuendo and the SSL Duende channel strip plugins (probably no compression though). When we’re ready with the mix, we’ll move to the desk and transfer the Duende settings to the desk’s analogue EQ and use all 24 channels to create an analogue sum (maybe using the Mixbuss compressor too). I’m looking forward to that. I’ll post sound examples and photos asap.

Share Button

← Previous post

Next post →

Leave a Reply

Your email address will not be published. Required fields are marked *