living sounds
Well-known member
It's funny to build this with a THAT20180. I replaced the old DBX2150's in all my console channels with 2181A to actually have less degradtion (or coloration).
the SSL driven sound I suspect is a result as mentioned above, many Op amp stanges and a couple of transformers...
Correct. I have worked on a few very early 4k's and the only transformer is in the mic pre....and even then its still a jensen!If I'm not totally confused there was only 1 transformer in the 1st micpre (not in the later version), the rest of the signal path is transformerless.
Yeah man. The first time I heard playback hitting the bus comp there was THAT sound (ha ha!! Literally THAT!). I pretty much immediately started digging in to the bus comp and decided on building the SB4000. I absolutely love that thing. Hands down one of the best additions to my setup.So what's the 4k sound? Maybe between 10 to 15 NE5534 / 5532 with some level up and down, coupled mostly with polarized electrolytic caps (back to back), some fet switches in between, some DBX VCAs (the old discrete ones, not the chip versions), a summing amp with a LM394 / NE5534 circuit? That's the basic signal path without EQ and channel dynamics, relatively clean. The eqs and channel dynamics have a very typical sound, together with the bus compressor in action, not unused. In my opinion the eqs and compressors contribute the biggest part of the characteristic sound. So I'd go for the bus compressor first, it is an easy, affordable and well documented project. Actually I did go for the bus compressor, it's always on my mix bus ;-)
So the "sound" of the SSL is really just a ton of 553x opamps and not-so-great VCAs, all fed from a massive linear PSU.
While I know it's not particularly cool to suggest software for such stuff here, I have to think that this is exactly the right domain for software to experiment and figure out if this is something you really want to pursue and exactly how much of this do you actually need? If you haven't, I would suggest you try that first. I don't know about this plug specifically, but Voxengo plugs are pretty nice.
https://www.voxengo.com/product/shinechilla/
This might do what you want without any fuss.
I sometimes use a parallel path with a high-pass filter into a distortion for some "HF Excitement" that I blend in to taste.
Funny that you mention software. I work mostly in audio post (editing) so most of my time is spent in the world of software. The reason I'm interested in making a hardware saturation device of some sort is because I want to see if it sounds different to software saturators. I've been looking at these (and other non-linear software processors) and I have a feeling that a lot of them don't handle aliasing very well (or at all). So I plan to put together an analog pre-mix hardware chain and test out my "theory".
Correct me if I'm wrong, but I believe pretty much all of these sat/dist software processors are performing a 2-8x oversample, executing the process, then down-sampling to get the output. As such, they should be free of any aliasing artifacts.
As for hardware, the Colour module is $100. Pick one up. While you're at it..Get the TM79 too.
That is true, and necessary. Contrary to analogue, where the intrinsic speed limitations take care of the issue, distortion in the digital domain extends the harmonic spectrum way beyond 20 kHz, which always challenges the anti-alias filters. Distortion algorithms often imply discontinuities, which are known to generate harmonic components up to digital infinity (Nyquist frequency).Correct me if I'm wrong, but I believe pretty much all of these sat/dist software processors are performing a 2-8x oversample, executing the process, then down-sampling to get the output.
They are not "free" of them, they handle them more or less well.As such, they should be free of any aliasing artifacts.
I agree that the EQs and dynamics (including channel gates) are the biggest part of the sound. However, lot's of mixers relied on driving the 4000 console mixbus a little into the red to get the characteristic "rock crunch". The 4000 starts to distort gradually wheras the 9000 series has more overall headroom, but once you reach the clipping amplitude, distortion sets in hard and sounds nasty.So what's the 4k sound? Maybe between 10 to 15 NE5534 / 5532 with some level up and down, coupled mostly with polarized electrolytic caps (back to back), some fet switches in between, some DBX VCAs (the old discrete ones, not the chip versions), a summing amp with a LM394 / NE5534 circuit? That's the basic signal path without EQ and channel dynamics, relatively clean. The eqs and channel dynamics have a very typical sound, together with the bus compressor in action, not unused. In my opinion the eqs and compressors contribute the biggest part of the characteristic sound. So I'd go for the bus compressor first, it is an easy, affordable and well documented project. Actually I did go for the bus compressor, it's always on my mix bus ;-)
Actually the opposite appears to be the case! There some very long and detailed threads over at GearSpace which examine various plug-ins in terms of their aliasing.
Testing Aliasing of Plugins (measurements) - Gearspace.com
Lets do it: The Ultimate Plugin Analysis Thread - Gearspace.com
The thing with the Color Modules is that they need a host of some sort which makes the whole exercise a bit too expensive!
And BTW, I'd probably go for the Royal Blue module if I was getting another one.
Cheers!
Depends on the source and the listening environment. To my ears converters these days pre-degrade the signal by using digital HP and LP filters that may measure well in many respects but mush up the presentation. And they are all Delta Sigma design, which never sounds as clear and punchy as multibit to my and many other people's ears (at least with real world clock implimentations)...As for SRC itself, in even multiples (x2, x4), I've found the impact to be negligible. Every mathematical function will impact and "degrade" the sound. Period. The goal of software developers should be to make that degradation minimal and in the event of artifacts, they should be the least objectionable at worst and even pleasant at best.
Actually many DSP engineers have little concern for things such as distortion and artefacts. They develop algorithms based on academic formulae, and very often they don't know how to measure teh results in a meaningful way. Asserting quality of a digital algorithm with another digital algorithm is often bound to let problems unearthed.The aliasing issue is a (presumably) well known one that has been recognized and (again, presumably) mitigated with oversampling for years. I guess I'm just a little surprised to see the topic show up again, and more surprised that developers are failing at implementing schemes for managing it.
Current DAW'suse polyphase filters for SRC. SRC is treated in the same way whatever the ratio.As for SRC itself, in even multiples (x2, x4), I've found the impact to be negligible. Every mathematical function will impact and "degrade" the sound. Period.
Often, their design brief is to provide as many functions as possible in the least possible dev time, sound quality being of s condary importance.The goal of software developers should be to make that degradation minimal and in the event of artifacts, they should be the least objectionable at worst and even pleasant at best.
I do too. The most revered digital audio specialists seem to agree that more than 64kHz SR is not necessary.For what it's worth, I run everything at 96k. It's been the best balance for me from a sonic/performance perspective.
I've seen people discuss the channel VCA's (for automation) on the 4K's as having some sonic imprint, but that was greatly reduced by the time the 9K's rolled around. I seem to recall accounts of engineers avoiding these VCA's because of the detrimental impact they had on the sound.
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