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I’ve always used pres in the audio interface I had at hand, some sounded better than the others. My favourite are Apogee and Merging pres because they are super transparent. If you want to get some colour on your vocals in post, you can always try Prex73 - it’s a free download.

@monomonoaudio thanks so much for responding I'll definitely check out pre73

You can adjust the envelope’s amplitude in the back panel section if you would like the envelope a bit sharper, I didn’t go into it too much. To be frank, both SH-101 I had at the studio sounded different to my ears, so it might be just a bit different envelope settings.

Love your taste in instruments! :-)

Would love to know how you feel about Mono One - if it meeds your expectations or not. Please don't hesitate to share your opinion. We have shootout between SH-101 and Mono One on our channel as well.

Thank you for your question. I've probably missed the VCO pitch wheel adjustment setting - I'll check it and include in the next update. Quantize option is a victim of how limited the Push 3 control UI is in a lot of cases. It's challenging and potentially opens the workflow to bugs, so I left it out. With regards to serial controls - my main concern is that it doesn't provide opportunity for any explanation on how it works. Unlike the Live version where you can at least read a bit, Push 3 might be even more confusing. Hope this explains it.

@@monomonoaudio Thanks for providing clarity, that all makes sense, in relation to the serial controls, I mapped these in your previous plugins using the Max object live.banks for Push 3SA control but was disappointed to see it was not even available as an option to map in Max in Mono One, maybe it could be something to consider adding as an option for more 'advanced' users to map themselves. I have to say it sounds and plays great btw, thanks for making it! It is great to see Max device offerings becoming more of a real replacement/alternative for 3rd party plugins and with the tight integration to Push 3SA this makes them even more appreciated.

@@lostreligion1313 ​ Gotcha. Let me look into this for the next update. I had to disable mapping for them for the Consolex8056 because when you use Consolex80 to scramble serials on all channels, it's reflected in Live's undo history, so it was a huge pain. I can probably roll it back for Mono One.

Thank you so much!

Thank you! I've also gotten MS-1 for reference, great value for money in comparison with the vintage one. :-)

Ah, this is so great! Yeah, I def see the point in having hardware, but it becomes so cumbersome at some point.

So happy to hear this and thank you so much for your purchase! Really appreciate it!

How CPU intensive is it?

This really means a lot to me! Thank you so much! Hopefully next devices will be more of use for you.

On my Mac Pro 7,1 (about 5 years old Intel Xeon something 8 cores) it runs at about 9% CPU for the first instance and about a percent for another instance. Live is a bit weird about CPU use, so it’s hard to benchmark it to be frank.

Thank you so much for your kind words! Very happy to hear you're enjoying those. I hate dealing with protected VSTs myself.

@ looking forward to stage and potential other devices.

If I remember correctly, some of the devices allow o change serial on Push, but I’ll have to come back to this as I plan to release some updates for Volume One in the next month or so. I’ll make sure to add it to push controls.

I’ll refactor the UI to make sure all controls are available in the next update. Thank you so much for bringing this up.

This is the new feature Live has added in 12.0. Check out the manual.

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.