I make videos about low-level nerdy audio stuff. You can email me at jason.zdora@gmail.com if you need direct help with anything pertaining to my videos.
@@__orangejulius You would want to get into your sound settings (the old one) by finding "more sound settings" in the new menu. Right-click your speaker icon and go to properties and change the bit depth and stuff there. Im not 100% sure what youre asking but thats the only place you could really manage any of this stuff. Just leave it at 32bit, honestly. Youre not hurting anything.
@@jasonzdora yes, that’s what I’m asking, sorry for not being clear. In the old sound settings, there’s no bit depth and sample rate option outside of “32-bit, 48k”. I was wondering if you had figured out a solution to that outside of using Voicemeeter because apparently UAD over Thunderbolt doesn’t have full WDM support, leading to issues with Discord and Zoom? I noticed minor clicks and pops during RU-vid and Spotify playback which led me down the bit depth rabbit hole.
Just when I was about to buy some of the paid M4L devices I found your video! Been using the free melda fx bundle but never realized it included an autogain compensator and more importantly with a sidechain workflow! Thanks for this really well explained video. You saved me a lot of time and some money :-)
You can also skip bouncing the "wet" signal to another track if you just go for the "File" > "Render" option. Helps speed things up a lot if you have something like the NDSP plugins and want to capture several mic permutations.
Interesting! So the squishy top end might be from the low frequencies affecting the overall waveform. Edit: you said Lowpass... hehe OK so the high end is just extra nonsense data and might be making it sound bad?
@jasonzdora Yes, exactly. All major lossy encoders use a lowpass filter because high frequencies use more data at the expense of the lower frequencies where we can actually here the best.
Noir Labs is awesome! Shortcut Buddy is in my default live set. Only issue, Volume Buddy requires Live Suite which a lot of people dont have. I think this one is accessible to everyone since its free. The only problem was that Melda never explained it properly so I made this video.
Did this in Cubase Artist 7. Cropping down the file, especially the beginning of it was the important part. Like many others I had tried this in the past but in a back to back comparison with the source there was an obvious difference in the sound, meaning the IRs +EQ I was trying to capture into a single IR was not captured properly. My final result was only close enough (maybe 98% perfect so I settled on that result). It probably comes down to how CA7 mixdown's (exports) audio from the track. It doesn't seem to do it perfectly because the new file is always a bit delayed. Had to line it up to the impulse I got from Libra (which has a perfect export of combine IRs) to make the EQd version of that start on time and not have excess silence at the beginning of the IR. Conclusion: Use Libra to combine IRs into 1 IR Only use this method to EQ said IR
Sorry your DAW isnt giving you exact results. I would try adding some silence before the 1-sample spike as a way of possibly mitigating issues with your DAWs handling of transients or audio file truncating. Not sure what the problem could be as I dont use Cubase. Glad you got decent results, though! Libra is cool but it has its own issues. Primarily the way it handles IR alignment. I dont like how they require you to manually align IRs from different makers. If you have an IR from Ownhammer, one from Bogren and another from York, you have 3 IRs that have 3 different starting points. Without the ability to visualize that, you are left guessing with their "Delay" knob. To me, the best way to blend IRs is Melda MCabinet in Edit mode, or Fractal Audio CabLab4. MCabinet will automatically align all IRs and allows you to add EQ, though its difficult to use. CabLab4 allows you to visualize the alignment which is a lot easier to manage than Libra. The only thing with CabLab4 is that you have to extract the IR through manual means as they only allow export to their proprietary format (so you do what I show in this video).
Since you said it's pretty much a bus, I'm curious if there's a way to adjust how much you're sending to it. Your routing works really well, but I ran into an issue when checking a mix that sounded like it was heavily clipping, but in reality wasn't. Switched it back to default and it was fine. Anyone else run into something similar?
The AUX is a bus, yes. You can adjust how much volume you are sending it by setting your send to pre-fader. Id have to see your exact setup to tell you whats going on but the first thing I would do is figure out if youre sending pre or post fader into the AUX. Then Id have to see where you are monitoring your signal and why youre hearing clipping. That could be a dozen different things.
If you set the speed too quick, yes. Its not meant to be used as a permanent effect. Its meant for doing A/B testing of effects chains and then disabling it once you get your gain staging correct.
It has the same latency as any other computer using Ableton and an Audient iD14 at 48khz with a 64 or 128 sample buffer. The imedance is also dependent on the audio interface. There is no difference.
IRs do not capture distortion, unfortunately. If that was the case, everybody would have already done that. They are "linear" so they can do things like EQ or Reverb but they cant do distortion.
For those who are wondering why is it like this. They have to license AAC and have to share revenue with the makers of the codec. This is one of the reason you will see AAA 192 in software like Bandicam and Nvidia ShadowPlay as well. The reason they can go up to 192 is because Microsoft has licenced it to 192 in such a way so everyone else can use it up to that. To keep your audio quality I suggest exporting with PCM. I am still paying greedy ADOBE just because they let me export on 512kbps AAC. I also bought Mirillis Action! instead of Bandicam exactly for same reason 512kbps recording.
Would love a Win11 decrapify video. I've done my own stuff here but there's always something I miss, esp with Win10/11. They're making it harder to turn stuff off, not like the old days for sure. I wish I could just run the core OS, that's all I ever need. Sure there's slipstreaming and whatnot, but lol effort. cheers
The best videos for this stuff are for gaming optimization. Im not the guy to make a video about that stuff because Im not interested in answering hundreds of comments for the next 8 years that could easily be answered by Google.
I noticed that exported IR's volume level needs to be increased around 6db's at least, somehow needs a normalize, how do we solve that? I got millions of other IR's and nearly most of them sound much higher in volume or does not need any volume boost.
You can normalize the IR manually with an audio editing program (audacity, your DAW, etc) or you can simply turn up the volume (the "Output" knob) on MFreeformEqualizer before exporting. If you have other IRs that are louder, its because they are probably normalized already. Drop them into Audacity or your DAW and check out their bit depth and volume levels.
It was a prototype but never got made, unfortunately. There were too many issues with the PCB and I didnt have the time to fix them. Ive kinda moved on since then and I havent re-visited this project since. Sorry, man! A good replacement would be a Tonographic Rusty Box and an EQ pedal. Sorry but thats the best advice I can offer at this point.
Id have to take a look at the recorded guitar you are trying to sample but in general, if there are any cymbals or other instruments in that recording, you are going to capture those extra high frequencies, resulting in a brighter IR.
I'm using isolated guitar tracks on RU-vid sampled into Pro Tools. Not sure why, but it's taking some tweaking after applying this method to get the tone closer. Maybe I've got a bogus plugin somewhere in the chain
@@PeachtreeGuitar What is your plugin chain? It sounds like youre doing a bit of an augmented workflow than what is in this video. Im just using MFreeformEqualizer to listen to the sample track. Then Im only using an amp sim for my DI.
That might be part of the problem. I had CLA guitars after Amplitube. Are you tone matching with all of the effects (reverbs, delays, mods, compression) in your amp sim, or is it dry when you tone match?
@@PeachtreeGuitar In this video, Im just tone matching with the recorded guitar I want to copy from. My side is just an amp sim and MFreeformEqualizer. No other FX.
Seem to have it set up the same way but can't get it to work. Does it have to be on a separate track? I've set it to Pre-Fx sidechain on the same track, AU version, might try the VST and see if it helps.
Not sure, man. I know Ableton will let you sidechain within a single channel so maybe take a look at which specific point in that channel you are referencing as your sidechain.
Hi Jason! Thank you for the video. I am using the same process in logic to record processed and unprocessed signals through my UAD x8p. I am here from the RME world and was wondering if there is a way to record more than 2 ST.AUX/4MONO AUX into your DAW through soft patching like in this video. Many thanks :)
I was looking at this for NAM!! Im so glad i found this video!! I only use a clean tone an overdrive reverb snd delay so im hope it can do some basic effects snd have my live rig. Even reaper delay would be fine for me. Any chance youve tried other effects with it?
Yes, I have! You are right that you would want to use stock plugins. But! If youre only using one NAM in mono, you can probably get away with more FX than you think! The big thing is learning how to manipulate the DAW session with a MIDI controller (a whole other huge rabbit hole).
@@jasonzdora that's awesome! I'm gonna need to offload some gear for this I think lol I've been using my laptop as my rig for a long time so getting reaper to accept my midi controller settings to some serious work lol Have you tried it with element vst loader or anything like that?
@@kyronnewbury Nah, I use Ableton. After a lot of testing, that DAW was surprisingly the lightest CPU-wise. Lighter than Cantable or Kushview. I demo'd Gig Performer and same thing... Ableton was lighter. Cool @ having MIDI. I suggest checking out MIDIkey2key. It lets you send MIDI notes to Windows and run scripts outside of Reaper. So you can do things like "wake up" the screen, start or stop programs, etc etc. Takes a bit of setup but its worth it. Also look into creating startup.bat files for your DAW. If you put them in a specific folder (windows startup folder) you basically just have to hit the power button on the micro PC and in a minute or so, everything is up and running without manual intervention. Takes a bit of tweaking but once you get it, its really nice.
@@jasonzdora that's awesome!! I also need to see if it will run Superior Drummer 3 for live electronic drums. The system requirements are Intel Core 2 Duo / AMD Athlon 64 X2 so it should work for it
@@kyronnewbury If youre talking about a live drummer on a MIDI kit, Id suggest using something lighter than SD3. Or at least pre-mixing the sounds so youre just triggering plain .wav files. If youre talking about a sequenced backing track, just record it onto a .wav file. Dont waste any resources. Thats more room for better guitar FX.
What does the audio file look like when you record it on the 2nd channel? If you dont see any audio information, have you tried playing ANY audio through your darkglass and re-recording it with the 2nd channel to make sure youve set that up correctly?
I did a quick hot glue and it seems to be holding. Theres a little ring that came with the lens that I was able to position right over the camera. Makes it easy to screw the fisheye on/off for packing. Honestly not bad! Ive found use for it, for sure. Just being able to watch people walk up without snapping my head around is great. Finding objects more easily, too.
Owning both, the UAD sounds so much more like the HW (that I owned 3 units for more than a decade). The Arousor causes that grainy highs that I hate from compressor plugins.
OK I'M GOING TO TYPE IN CAPS BECAUSE I'M SO FUCKING EXCITED ABOUT THIS, I'VE BEEN DEALING WITH THIS PROBLEM FOR 15 FUCKING YEARS, HAVE NEVER FOUND A GREAT SOLUTION, SEARCHED HIGH AND LOW, FOR YEARS, NEVER. WAS DEALING WITH THE PROBLEM AGAIN TONIGHT AND WAS FUCKING PISSED AND FOR SHITS WENT ON ANOTHER SEARCH. HAD BEEN DOING RESEARCH ON THE UAD C-SUITE FOR A WHILE BUT ITS FUCKING EXPENSIVE AND WOULD HAVE TO BUY A FUCKING APOLLO INTERFACE, WHICH I WAS ABOUT TO DO, JUST DROP LIKE A THOUSAND AND BITE THE BULLET. I DON'T KNOW HOW I MISSED THIS PLUGIN. IT'S FUCKING PERFECT. YOU'RE FUCKING PERFECT. HOW IS THIS FREE? I PAID THE MAN FOR IT BUT FUCK. HOLY SHIT. THIS IS WHAT I'VE BEEN SEARCHING FOR MY ENTIRE LIFE. I'VE FOUND THE OASIS. THIS IS EVERYTHING. YOU ARE EVERYTHING. FUCK. I LOVE EVERYTHING AGAIN. FOOD IS TASTING BETTER. SMELLS ARE MORE VIBRANT. MY LIFE IS FINALLY COMPLETE.
I dont know, brother. Sorry. But I think I saw another video where a guy used a "capture card" and the right cables with the right connections to connect between his RC2 and his iPad.
Here I am, in my own comments, pointing out that I had to re-watch this video so I could do this setup on a new rig. Im so glad I make RU-vid videos about really specific things like this. Thank you, Jason from 2ish years ago!
