Recording a modular into a sequencer …what is the best intermediate device ? Is a standard mixer the best option ? if so how would EQ be set, should anything else be used ( pre-amp etc)? What’s a good method?
Depending on how many outputs…optimally you’ll want a direct box into a mic preamp… EQ/other processing to taste, but I there is no rule on how EQ should be set…hopefully you’ve got all the sound you want in the modular already without it?
Are you recording the audio output into a mixer/recorder or a sequencer? I believe that a sequencer will only record note data (not audio) via CV or midi. Note data includes pitch (note number if by midi) and duration. Sequencer output needs to go back into your modular or a compatible synth, again via CV or midi.
What sequencer do you have?
I think he means a DAW sequencer like Logic or Pro Tools. You need a simple AD converter or interface, like an apogee duet or apogee one. Or computers have their own sound card usually with line or mic inputs, which should get you started without an interface. But interfaces act like sound cards and have usually higher quality components and mic preamps. Synths can be anywhere from low level signal like guitar outputs to line level, which is usually +4 db for pro gear or -10 db for consumer gear.
You best bet for a single output synth is something like the apogee one, but it only has one preamp, so one one source at a time can be recorded. I think it switches from line or mic input so you should be covered as far as level, you just need to test it out to get the optimum level.
FWIW, I pass all my analog synths through an analog mixer, with some external effects that I switch on or off for each input via the send/receive, then it’s straight to my audio line in of my Mac. No special audio card, or external digital device.
I set the input sampling rate in my Mac to 96KHz @ 32 bits resolution, and work in a multi-track recording software, and set the final mixdown to 44KHz @ 16bits for publication.
The effects I sometimes/always use are: Delay, Reverb, Chorus, Flanger, Phaser. I never had to EQ anything yet, but it could be useful in some situations.
My two cents on it.
You have a 32 bit interface? What is it?
It’s simply the integrated sound chip in my Mac Mini. The only thing it can’t do is full duplex. But most of the recording software can do this anyway.
Thanks for replies…yes I’m talking about recording into Sonar. At the moment the modular goes into the mixer connected to PC via EMU sound card.
I was wondering if anyone used pre effects or even compression in the chain from the modular. ? Perhaps there were some “methods” used in 60’s recordings etc?
+1 with thealien666. No need to run through a DI or mic-pre. In some cases if you happen to have some badass tube mic-pre, run it through it for some effects, but for the most part there is no need. The best thing you can do is record everything at the highest possible sample rate you can. 96k @ 32bit is ideal, but your processor may suffer if it’s slow. From what I have seen, the majority of audio interfaces on the market these days can do 96k & 32bit. Personally, I’m using a Focusrite Saffire Pro 40. As for eq, the only thing I really mess with is around 500Hz. Taking a bit of that out can help tighten things up a tad. Also, I like to remove the llow end on anything that is not intended for bass tones. Helps clear up the mud.
Hmm interesting info thanx! 
~ if you are going to use a preamp I absolutely love my FireFace 800 — and if you want to add more channels to it, the OctoPre MKii Dynamic FocusRite is a particularly cool little unit with compression available on each channel, and those dual XLR or 1/4" inputs. I DO agree about recording direct into a preamp at least if not direct into the computer 
I don’t use XLRs for analog keyboards and synths very often unless I am recording a very special live situation with acoustic performers at the same time, but even then I would most likely go direct into the patchbay from analog hardware outs.
I have been reading a little about 32 bit recording, because I know that there is no advantage to 32 bit over 24 bit, because 24 bit already has a larger dynamic range than any analog gear. It seems that 32 bit is only used internally in computers to increase headroom for digital processors like eq and effects, and 32 bit ad converters are not common, even in mastering studios.
Muksys, could you give an example of a 32 bit interface, because I couldn’t find one.
And as far as higher sampling rates, ie 96, 192 khz, I have tried these out, and there is very little difference in sound. Some converters have different electronics for the different sample rates, which can change the sound slightly, but whether its better sounding or not is not set in stone.
I wouldn’t worry about compression with synths, basically the dynamic range is going to be determined by the envelopes anyway. As long as its not clipping and you have the right levels it should be fine. I am not familiar with your emu interface, but it sounds a little outdated. There are plenty of cheap interfaces out there today that are pretty much as good as high end studios are using, as converters have pretty much homogenized as the tech has advanced.
Unfiltered37, when I am recording I always set the bit rate in the DAW (Cubase5) to 32bit (float). I did a bit of reading right quick and it seems it doesn’t matter as far as the interface goes because the computer is doing the recording. I guess the several interfaces I’ve purchased I only paid attention to the sample rate, but always have had the option of 32 bit float in my DAW, so I assumed it was the interface. Guess I was wrong. It’s more a DAW thing rather than an interface thing. But, with that said, when we had out EP mastered I was talking with the engineer about recording at 96k and I could not hear an audible difference. The way he explained it is the higher the sample rate the more data there is for him to work with. He said its much easier for him to master higher sampled recordings as there is more dynamic content. He was explaining that something recorded at 44.1 or 48 is harder to fix trouble areas with EQing and compression as there is not a lot of data available to work with.
I agree regarding compression. I’ve run into situations where compressing a synth actually made it worse because it got rid of the dynamic percussiveness of the particular patch. The only time I work on compressing a synth is if there are a lot of high resonant filter sweeps, especially on sawtooth waveforms.
Yeah, higher sample rates can be useful for things like flex time and processing and such. Also evidently a lot of DAW’s convert bitrate and sample rates to higher values for internal headroom and processing. I guess its like oversampling, when you insert extra samples in order to use a gentler alias filter. I have not seen any interfaces or converters that did more than 24 bit, and I know they would exist if they had any benefit.
Yeah it’s crazy with analog synths, they defy the need for studio processing. They create their own dynamic range and spectral content, and the constant moving filtering is EQing itself, so in order for the processors to work, the patches basically have to dull and static. But you can get great results by just running though tube compressors or whatever in bypass mode or with slight limiting.
A limiter is one of those necessary devices when recording a modular. If it is not built into the sound capture device you are using. Nothing worse than recording a great performance only to find the levels were too high and there was digital clipping. The volume on a modular can vary greatly when experimenting, so getting into the RED zone is easy without a limiter. Using a WIN95/98 Lexicon core32 with a built in soft knee limiter has saved many a recording where a few peak levels would have clipped.
The truth of the matter is, that @44.1KHz sampling rate with 16 bits quantization resolution, it’s already more than adequate for the human ear. It already provides a dynamic range of 96 db, and a 1 to 22 KHz frequency response, which is already better than the hearing of a newborn baby with perfect ears.
Just as a reminder, a 96 db dynamic range equates to the difference in sound level between a pin dropped on the floor, and a pneumatic jackhammer at a distance of 1 meter (3 feet) !!
And, as stated by unfiltered37, audio recording software use 32 bits at higher than 44.1KHz re-sampling to raise the headroom, reduce to nil the aliasing and quantization artifacts, and lower the noise floor when performing calculations for effects and such.
Those “purists” that claim to be able to perceive an audible difference between two recordings of identical audio material having been sampled at 44.1KHz @ 16 bits, and 96KHz @32 bits are misleading themselves.
They might as well have pointed ears like Spock. I would like to dare them in a blind test. As did one of my friends who was convinced of this, but was astounded to find out, much to his dismay, that he finally could not tell them appart.
A Minimoog D, being played by someone and properly recorded using digital means, cannot be distinguished from the real thing playing the same thing live. Period.
NOTE: I’m not talking about individual notes being sampled off a Mini, and being played back on a sampler/rompler. No, that’s a completely different (really bad) thing. I’m talking about a complete musical performance being played by someone and fully recorded in it’s entirety.
Always thought the 24 bit recording helped when mixing many tracks together. While a single live stereo recording may not have a noticable difference, I would imagine mixing 20 or so tracks with 16 bit versus 24 bit would have some sonic difference, no?
thealien666, I agree. Matteroffact, when I was speaking to the Mastering Engineer, I told him that I have recorded at all available rates on my interface and did not notice a difference. He said this is true, but the higher the sample rate, the more information there is available to the mastering engineer to work with. Granted, all audio CD’s are at 44,1 @ 16bit, but it’s a pretty safe bet that the audio was recorded at at least 48k @ 24bit. My rule is if the option to record at a high sample/bit rate is available, do it. It can’t hurt. I do know that I started working on an album that was originally just messing around and started the session at 44,1 @ 24bit. When I decided to work on more tracks and develop a solid album from it, I decided to keep everything at the same rates. The recordings sound good and I’m happy with them, but when I listen to them next to our bands EP which was recorded at 96k, there is a noticable difference. Everything fits together better. There seems to be more “space”. I’d say it should be environmental differences, but it’s all recording using the exact same gear. At any rate, some of the best albums I bought from my youth were recorded in garages on shitty cassette multi-tracker, or some dudes living room converted into a recording studio ala Albini, with shitty gear. If you need to scientifically calculate formulas to recording your music, then you’re doing it completely wrong and should toss the towel in. Just set some stuff up and hit record! Strive to make musically awesome recordings, not awesomely recorded crap.
Yes CZ Rider, I was talking about a final rendering.
Because of course, as I stated earlier, it helps greatly to record master tracks at higher sampling rates and higher resolution to eventually mix down those tracks with added effects. That’s why I use 96KHz @ 32 bits, in order to have the highest resolution possible before downsampling for a final mixdown.
Because it’s always easier, and better, to cutdown extraneous information than to approximate it with oversampling when it’s missing.
word
Yes muksys, one only has to notice the poor audio quality of some of Led Zeppelin’s albums (Physical Graffiti comes to mind) to realize that it’s not the container that’s always important, but the music it contains.