Glad I was able to clear this out in the video! Appreciate your kind words!

Thank you for your question. The shelf is around 2 KHz, but the bandwidth is wide, so it captures frequencies up to 1 KHz. If you low pass around 260 Hz and cut around 6-8 dB on the high shelf, most compression will be triggered by the mid range.

@@monomonoaudio Awesome, thanks. Will be interesting to test out. I've been deeply invested in mapping compressor threshold amounts to velocities via Ableton's Expression Control or manually automating the threshold for audio, allowing gain reduction to be consistent from soft to loud. Full dynamic range compression in small GR amounts, while SC filters are tuned to focus on specific frequencies depending on the key of an instrument/song. But I digress, thanks for making a great compressor with a lot of character.

Wow, never heard about any technique like that. Very interesting. If you happen to make a video on this, I’d love to check out. Thank you for doing some awesome stuff with devices, really. Glad you found them useful.

wow! the entire product is on another level. it was not boring for me. it was a very nice showcase. the remaster from the track is great as well. i purchase it now.

Thank you so much for watching! Appreciate your generous comments and if you got any questions, don’t hesitate to ask.

Thank you so much for your suggestion. During the tests, we felt the link function, which is available on other devices, did not provide any value, but rather still required to unlink and adjust the input on the Pre - which took more time. And on top of that in most cases compensation would still be required on the EQ output and since they are split, it wasn’t a straightforward feature. Once you’ve learned the devices, gain staging kinda clicked after a few uses. Hope this explains our thinking and I hope I understood your suggestion correctly.

Hey! Thank you for your question. It really depends on my goals: if you're not strapped on CPU and you enjoy how it sounds, I'd add it on all groups and Main bus. Even if it cuts into high frequencies a bit too much during the mix, you can compensate it during the mastering. In the mix I prefer to leave some headroom in high frequencies anyway. If you want to conserve some CPU - just drop it on your Main bus: either as a clipper - putting it as the last device with Soft Clip function on, or you can just use it for character early on the Main bus and put your mastering chain on top. Hope this makes sense and answers your question. Don't hesitate to ask.

@@monomonoaudio no this makes perfect sense. I'll be purchasing this weekend. Thanks for doing this. I was actually talking with a friend about making something similar but you beat us to the punch and honesty looks like you put a lot of thought into it to achieve a really really cool final result.

Appreciate it a lot! I think you guys shouldn’t be discouraged by any available products similar to what you’re thinking. Especially when you see a clear value and improvement you bring to the table. If anything it might be a sign of actual demand! So let this release not stop you, but rather inspire with new ideas! Looking forward to seeing what you got in the future.

Great question, thank you so much! I tried not to go into deep explanation of the gain staging in the video, cause it can get very technical and very long, so I appreciate this opportunity to talk about levels a bit more. Short answer: the saturation curves are tuned to audibly saturate at about - 11 dBFS (Vintage mode) and -5 dBFS (Modern mode) and clip around +9 dBFS. This is mostly done to provide convenient interfacing with stock plugins and stock devices as in many of them the starting points are 0 dBFS (threshold of the stock compressors for example). This is also done to have a bigger margin for error if you have a little limiting understanding of gain staging. I think most digital "analog" gear is trying to match hardware levels (assuming +4 dBu - nominal studio level in analog domain - 0 dBu equals ~-20 dBFS - -16 .. -12 dBFS essentially matches +4 dBu in hardware), but I thought a better approach would be to match the saturation to the production workflow with stock plugins and how they are designed to interface with producers. Long intro: If you consider how the analog gear works, it all has various dynamic ranges - basically safe levels between the noise floor and level where signals start to distort. In analog recording all components of the signal chain - preamp, eq, console summing amps, insert or output section, going into outboard, then onto a tape - will have different dynamic ranges and subsequently provide different headrooms. So what gain staging in analog environment means, is that engineer has to adjust gain at each stage of the signal chain to maximise the loudness of the recording while importing as little artefacts (distortion) as possible. Unless artefact is an artistic choice of course. Nominal level for professional studio hardware is usually +4 dBu (which roughly and generally matches -16 dBFS in digital domain). With introduction of better components, better tape formulas it actually shifted higher and some engineers in studios even adjusted the 0 position on the VU meters. With 32-bit processing the situation changed considerably to +4 dBu became 0 dBFS in a way. People generally (and I want to emphasise generally here) tend to aim for a signal level slightly below 0 dBFS on the master bus now. That's because all interfaces have their own max output, so for whatever the interface you've got, 0 dBFS will basically mean maximum level for the interface to safely pass from the digital domain into an analog domain (speakers). In production in a lot of cases, samples you get from various sample libraries are normalised - meaning, they usually hit around 0 dBFS on peaks. Modern plugins and compressors which are not based on the hardware also tend to have 0 dBFS starting points as well. Without going into too much technical details, due to 64-bit processing inside Ableton Live, you can pass the signals between your devices, between plugins and tracks safely even if they are +40 dBFS (due to 64-bit processing) - so you can have signals way above 0 dBFS inside Live (before they hit Main out). The way I structured the saturation is that when your signal hits 0 dBFS - it hits the upper range of the sweet spot (at least for me) anything below 0 dBFS will sound gradually cleaner, anything above that, will step into distortion territory fairly quickly. The exception is the Consolex80 device, which unlike the rest of the devices can actually act like a clipper which soft clips the signal to slightly below 0 dBFS. So if you mix and master yourself, you can clip it on the Main bus and push your tracks with track faders in or out to dial the overall saturation. In my mastering process, I tend not to use a lot of compression (1-2 dB of gain reduction at most! if any) and instead I use soft clipping to hit the levels I need. In this mixing video I 1. wanted to match the level of original mix, so I've used Optocomp to make the mix louder and 2. I've used Tape to clip the signal in the end. That's why I've put out the soft clipping on Consolex80 in the end of the video and let the Tape do the work. Hope this makes sense and answers your question. And pardon my rambling. Don't hesitate to ask further questions and voice your concerns.

@@monomonoaudio Your answer to this question is a pearl of wisdom. Thank you for sharing the thinking behind your calibration decisions. You have made the equipment foolproof, so to speak, by considering the most likely use cases for music producers and "modern" music making. I'm glad to know this and will be happy to use this approach.

Wow, thank you so much! I really appreciate your interest and I wish I had time to take any mixing and mastering work, but I’ve stopped doing any commercial work for a time being, cause it is just very difficult to make commitments and deliver on time. I also have so much exciting stuff I’m working on and very excited to release. Hopefully at some point it will be possible!

@@monomonoaudio hey keep dropping max devices like these and you might be putting some ME’s out of business anyways haha excited to grab this next batch 🤘🏼

Thank you for your question. It's not IR based, but a combination of multiple filters and transfer functions.

Thank you so much! Glad it was enjoyable. Not much long to wait! :-)

Thank you so much for your comment, appreciate your feedback. Apologies for the inconvenience, this is the first time I'm doing anything like this and my editing skills are too limited at this point. Hope you're able to see something useful from the video. The devices are out on July 1 and more information will be posted then. If you're curious to look at them up close and see the comparison with hardware, please feel free to jump to our instagram page to check it out.

@@monomonoaudio No editing necessary. Just record your videos (in the future...) using higher scaling. Just check Ned's own stuff - it's all nice and easy to clearly see :)

@@Artek604 Will do my best. Thank you!

Thank you so much, appreciate it! Yeah, the mix came out alright, I think. I should have probably kept it a bit more dynamic, just a couple of dB less pushed into the mastering chain, but otherwise, it's alright.

Missed the comment. The new set of devices is available starting July 1

There are Max Devices from monomono.audio

It's based on various Studer and Revox tape decks I've had experience with, but with slightly exaggerated character - essentially from the 90's to 60's. You can check more information and hear examples on our website.

I second this 🙏🏽

Thank you so much! I feel it's going to be extremely difficult to beat fabFilter Pro-L 2 - it's just so good when you want some limiting done. But in all honesty I stopped using any limiting in the first place - or very gentle limiting (< 1 dB of GR).

@@monomonoaudio tbh I'm finding that more on more, just your EQ and tape does so much heavy lifting in recent tracks (process of being signed so thank you!). That said, while I'm happy with using Airwindows console stuff, a monomono preamp would be killer (and would throw potentially 3 plugins in the trash)

Happy to talk about those, best money ever spent for sure.

That's a good question. While working on this, I did not think there's enough control over the envelope to be able to use something else in the sidechain. I still don't if I'm honest. With external sidechain I'd personally prefer control over attack and release, while content dependant nature of this compressor is limiting in this regard. I would probably also prefer a different UI to be able to make decisions a bit more reliably. I don't think I know any content dependant compressor with external sidechain input. Is there? What would you send into the sidechain instead of the original signal?

Ah, I blanked there for a second - the simplest usecase would be processing the signal before sending into the sidechain, which makes a lot of sense. I personally very rarely used anything apart from high pass on the sidechain in my mixing and mastering work - that's something of a must for me, so I've added that. I guess external sidechain would be a good suggestion for v1.1 - appreciate you bringing this up.

Definitely not the last set of plugins. I'll post new videos soon to show off what we've got in the works.

Please keep us posted! @@monomonoaudio

It's not available yet, it's one of the first plugins we've made and it needs to mature to become available to everyone. But it's definitely coming.

Thank you for your purchase and suggestion. I'm really slow on populating any video content, unfortunately, but I'll get to it. :-) Short answer: I would say Glue and Optocomp are quite different not just in operation, but in sound as well. They both have a place in the toolkit, in my opinion Optocomp has more of a leveling compressor sound to it. The compression is quite transparent even at high gain reduction values but can be quite characterful if you drive it hard. Glue is more versatile with dedicated controls obviously based on SSL bus compressor. Detailed: Optocomp is inspired by the "optical" compressors (like LA-2A and LA-4 which we've used to model the response of the Optocomp) - with 2 main controls you can dial the amount of desired compression. We wanted to bring the experience of working with hardware into this device - where you just don't have to think much about the settings, just ride it with your material and see what you like. Input dials the amplitude of the signal which hits the gain computer and calculates gain reduction amount. Basically, you push more with input if you want more compression to happen and roll back to get less dynamic reduction. You compensate the dynamic changes of the output with Makeup dial. Based on the VU meters you can compare the input signal with the output to match them if preferred. Like it's hardware counterparts (and I assume Glue compressor as well since it's based on the SSL Bus Comp), it's a feedback compressor which feeds the compressed signal back into the sidechain and calculates based on that. Attack and release are content dependant - meaning they are adjusted based on the gain reduction values. Threshold is fixed at around -20 dBFS, but the GR curves are very soft. Which makes it quite transparent even with a lot of gain reduction and it's quite musical as well. It also includes tube/transformer output which softly clips the compressed signal and saturates the signal with even and odd harmonics. I think you either will like the sound of Optocomp and will just breeze through any dynamic issues you have once you learn it a little bit, or you might never use it again. :-) I'd suggest to experiment with different gain reduction ranges, see what you like best, what works best for your material. If you want to bring up the quiet parts - crank that input gain and lower the makeup to match the incoming signal. Try mixing it with original signal with Blend control. If you'd like to control the peaks a bit - hit the compressor just slightly to get 1-2 dB of reduction and boost the makeup a bit. Overall, we designed it be easy to experiment with. We tuned the GR curves and envelope response to what we felt makes it a very balanced tool. I ended up using it on basically everything - individual tracks, vocals, busses and master. Hope you'll enjoy working with it and don't hesitate to let us know if you have any feedback. Thanks, Nikita

@@monomonoaudioThanks for the detailed response. I purchased a UAD mix bundle a couple days ago, after using your eq, comp, & tape… i wish i hadn’t purchased that bundle… it’s a bargain what you guys packaged these for. I use tape on EVERYTHING … I compare it to the UAD studer tape and i can honestly say your device is better. I’ve been messing around with dialing in the optocomp as well on multiple tracks. I’m only having trouble getting curves to sound good. Maybe i just don’t understand how to use a pultec style eq yet, i get the controls but i don’t hear anything besides boosts in volume on certain frequencies? than again i’m new to mixing and paying attention to these details. When I purchased your bundle it says “volume one” I can only expect and be excited for your next batch of plugins with “volume two” thank you for creating these.

Thank you so much for such a glowing review! Really warms my heart to hear it's valuable. I think I've missed a few crucial details in the Curves EQ to be honest. The update is coming in the next few weeks which I think will bring it much closer to what I expect passive eq should sound like and how bands interact with each other. We've carefully looked at the circuits with our hardware engineer and I'm very excited for the next update. Unfortunately this will probably render the existing sessions differently with the same settings, but you don't have to update device in them. I'll post an update to this channel once it's released and I'm sure Isotonic will send out an email as well. In general, I think EQP-1A3 is the most effective and one of the most beautiful sounding EQs ever. It's so easy to dial just the right sound, it's absolutely invaluable. It takes a bit of time to getting used to how bands interact, cause all bands basically affect each other, but once you learn a it a little, once you get used to having it on your buss - it's just so quick and simple and it sounds good basically on anything. It also saturates nicely, just a bit to bring that tube crunch to it... I should retreat before I salivate too much. Again, thank you for your purchase, stay tuned for the update to Curves and don't hesitate to let us know if you need any help. Thanks, Nikita

Oh, forgot to address your "Volume One" question: we're definitely not stopping with just these three. We have plans to release new devices around April-May next year.

@@monomonoaudio Weird request, but would you be able to make a custom TAPE for me? I love the fact that i tap the reel and it makes the pitch drop, i’m wondering if it’s possible at a price to get a custom “Tape” max for live plugin where the pitch drop lasts as long as it’s held for like a real “reel to reel”? Same interface and sound of your tape plugin just that small custom adjustment? Let me know if that’s possible! Not sure if you do custom devices or not.

Thank you so much for your purchase and appreciate you reaching out. I'm assuming you're talking about Curves EQ? If so, there should be a second page of mappings called Additional which provides access to the gain staging and hpf. You should be able to access mapping page selection if you press on Curves EQ in the top row of buttons above the screen. Do you think you could check if you have that? Attaching screenshot how it should look like: www.dropbox.com/scl/fi/kh07kypa0p42938fdom9r/curves-page2.jpg?rlkey=8s5vatcpywaiwstg32zng0kmq Let me know if you're talking about something else.

Hi, thanks you for answering so quick :-) I do have a second page. If I pass on the Curves EQ there is just one label ”Quick“ with the same parameters like before. No ”Additional“ next to it. Strange. I'll install it again. Let's see...@@monomonoaudio

@@nilssanders3165 Thank you and please, keep me posted if it fails. I'm a bit puzzled why this could even be an issue. My Push 3 is 1.1.12 at the time of testing.

sadly it do not work. where did you put it on the push? I'm on 1.1.20b4@@monomonoaudio

@@nilssanders3165 So sorry to see you experiencing issue with this. It's very hard to say what exactly could be the issue. The screenshot is from the Push 3 in Standalone mode, but I have not installed any beta updates. I'll test it with the beta hardware when I get to it and see if this is an issue beta maybe. Could you please send us a direct email to [email protected] and I will send you the latest build? Otherwise, we're planning an update in December to address some of the other issues we've noticed, we'll investigate more on this.

Thank you for asking! You will be able to stumble around them after October 24 when they are released at Isotonic Studios website.

Thank you so much for your interest. It's coming out on Isotonic Studios isotonikstudios.com/ in mid-October.

@@monomonoaudio thanks. Maybe just put that kind of information into the info box below the video for marketing reasons ;)

@@distorsonGreat point, did just that. Thank you! 👍