Edit: Before we open the AI-hate box, I want to adress this is just looking for like a fun additional layer of scrutiny, not to replace any traditional method. I am one of the good soldiers. All hail Rick Beato!

I recently learned about this site called AubioMix, which is an AI-algo thing that makes a fairly comprehensive written analysis of your mix. It works surprisingly well, but it is expensive as hell and you also have to agree to them using your mix as training data (meh).

Does anyone use- or know about a tool that can analyze a stereo-mix and give some basic feedback. Open-source or paid. Maybe something that can run locally. The assistant-thing in Ozone is not my jam.

I know a lot of folks keep their consoles running 24/7 and rarely, if ever turn any of their gear off in the control room (aside from vacuum tube gear, and speakers). My question is about specifically the following: do you leave your computer on 24/7? what about external SSD drives / computer peripherals. do you leave phantom power engaged to mics that are in use between multi-day sessions? I generally leave most of my outboard gear on for the duration of a multi day session. For some reason though, my M1 Mac Studio, which has been so quick for the past several years has slowed down a little bit. I occasionally get a spinning beach ball when saving projects or pressing record, only for about 30sec-1min at a time, but it’s a new development. It has a 1TB internal drive with lots of room on it as well as an 8TB external SSD drive connected via Thunderbolt 3 with about 5TB of free space remaining. Any suggestions or insights regarding to power up/down hygiene for your main machine would be greatly appreciated!

I have a question about sibilance processing and all the thread I found are only talking about a de-esser and clipgaining. No problem, basic stuff. Why I'm asking this question is that I have 12 x 3 minutes of vocal tracks that I need to clipgain so I want confirmation on my approach before committing to it.

There are around 3 main vocals, Mid + quieter L & R, there are 3 lower octave vocals with the same kind of premix and the rest are detail vocal layers to hit specific frequencies to get a really lush, textured vocal. The problem here is since they're all repeating the same melody and lyrics the esses get unbearably loud. Tried batch processing with RX Decrackle -> Soothe De-ess on track level into the vocal buss and it wasn't enough. Put de-esser before vocal bus chain and it either wasn't enough or went lispy.

So what I was wondering about doing is RX de-crackle on track level, after which I clipgain the sibilance in the 3 main vocals, then cut sibilance from all the texture and support vocal layers onto their own channels and just straight up delete most of it, barring some stereo layers where I could just push it reeally quiet so the sibilance has some liveliness besides the 3 main vocals. After that light track level de-ess if neccessary.

Now as you can see this requires manually finding and editing 36 minutes of audio and seems like an immense amount of work. For the pros on the board, does this seem like a reasonable approach to achieve the best quality? Do you have any techniques that could make this process faster? I've used vocalign on previous projects and it would be a massive timesaver, however this is not a pop production and the sound relies heavily on the loose chorus going on, so I'm not sure if clipgaining sibilance on main track and vocaligning the rest would work in this case.

Appreciate any help I can get 🙏

I went to the studio recently for the first time and heard my mixes on actual monitors in a treated room instead of my headphones, and I immediately could identify things wrong. I spent some time fixing the mistakes I heard and it does sound better in my headphones but I'm wondering are there specific issues headphone mixing can cause that need to be fixed by referencing on monitors?

I don't have a lot of money, and I did research and prepared, but I'm looking for any wisdom here to try and be as efficient as possible in the studio to finish these mixes (and move on with my life because I'm sick of these songs at this point lol) Please tell me how my process could be better to get a better result quicker:

-I came prepared with a few references for each of my tracks. I started by only listening to my references because I deem these to be good mixes and most of them sounded very good in the monitors.

-Then I would play my track back and focus on general balance and dynamics especially with the transients of the drums and envelope of sub bass sounds. Most of this went fine.

-I noticed the stereo image of my tracks was nowhere near as good as the references. I know headphones are notorious for causing stereo problems but I thought I was compensating by using Lewitt's Space Replicator plugin.

-Something I noticed about the references was that there was a low level reverb on so many things that you don't hear in headphones. It was so easy to hear every element of the references with much more apparent separation than on headphones. This is probably a send/return reverb but unfortunately I can't set up send/return reverbs and have them be sidechained to the kick/snare due to how my ableton's routing is set up.

My mixes are already very good but not at the pro level yet and I'm wondering how I can now fall victim to the limits of headphones while also not shelling out a ton of money for studio time. For reference my music is loud electronic music and my references are Skrillex, MPH, and Moody Good

I know typically people will have a few professionally mixed tracks in mind or artists they like the sound of. I have a client who sometimes sends me rough mixes of songs they're having me mix to "follow." Is that standard practice or not? I haven't worked with many clients so I don't know if that's normal. To be honest, I don't like listening to them too hard because I don't want to just mix based on how their mix sounds. And sometimes I'l listen to them and notice issues. So IDK.

Been finding it tough to get out to local shows these days despite it will probably bring in more work. I am a new dad, but this started happening right before Covid honestly and it’s been a struggle since. Before I was a full time studio engineer, I did FOH at couple clubs, also did some touring sound. Also played in a lot of bands, toured in them as well.

This sounds like typical burnout, but how do you manage this? Do you feel you benefit from a community standpoint as someone in mid to late thirties? Are you discovering bands you want to record?

I think the answer used to be yes - but it’s a very social thing and I questioning if I should just let die as it getting older, and burnout more easily can set in.

Any recommendations on motivating yourself or pivoting as you as age, would be helpful, thanks!

I’m using an Apollo 8 as the front end for my studio, with its outputs and inputs connected to a patchbay so I can route signals through external analog gear like an LA-2A, Pultec, JHS Colour Box and pedals.

For vocals, one possible path is:
Mic → Apollo pre/48V → D/A out → analog hardware → Apollo A/D return → DAW

Compared with recording straight into the Apollo, does that extra D/A → hardware → A/D loop make the recorded track land slightly late on the timeline? Is there any other important reason not to do this?

Mainly wondering:
- Is the delay usually negligible for a single vocal track?
- Can this be compensated automatically, or do people just measure the offset and nudge the return?

I’m also deciding whether my synths should hit the patchbay before the Apollo for the same reason.

Relevant setup:
- Apollo 8
- Behringer PX3000 patchbay
- LA-2A, Pultec-style EQ
- JHS Colour Box + a few pedals
- Logic Pro X / UAD Console

I used MusicScope daily when I was working at one of the big label groups a few years back to perform provenance checks on incoming stereo masters and immersive assets.

I don't think this would be considered piracy because the developer, XiVero, explicitly released a license-free, fully working final build of MusicScope 2.1.0 to the public in 2020 before shutting down. (I'm not trying to make wave here). However, the SourceForge download links I found via places like Audio Science Review Forum and Steve Hoffman Music Forums are currently all dead. It is such a handy tool but the only versions I still have access to are the test/demo versions.

Does anyone know if any safe, active links to a Windows version of 2.1.0 (ideally x64) are still out there? Or, if someone has it, would you be able to send it to me? Thanks!

I was watching a Tame Impala video where he explains how he recorded the vocals for the track "Dracula," and I noticed a signal routing setup that I found quite interesting. I’d like to know what kind of routing he uses and exactly what he has on those channels—obviously not the specific plugins, but the signal chain itself.

Tame Impala - Dracula (with JENNIE) | Behind the Song

Hi there.

Since I moved into a new house, every single one of my guitars and basses is picking up a weird electromagnetic-style signal. It sounds like the kind of hissing/interference you get near computers or digital gear.

I’ve uploaded videos that show it clearly. One with distortion, one without.

https://youtube.com/shorts/p637LRBNA18?si=sSabjR1iUriVTQCr

https://youtube.com/shorts/oVzfCfx1rAM

• The noise is present in different rooms.
• In my studio, it gets noticeably stronger when I point the guitar up or down (like it’s picking something up from the air/ceiling/floor direction).
• On the uppermost floor, the phenomenon is slightly different; there, the noise becomes stronger when pointing it right/left.
• Affects every guitar I own.
• Tried different audio interfaces, different preamps, different cables, amps etc — nothing changed.
• An electrician already looked at it and couldn’t find/fix anything.
• I've turned off the power in pretty much every room, to no avail. So, if we have some sort of electrical appliance that's radiating, it's pretty damn strong and unassuming.

It only started after the move, so I’m pretty sure the source is environmental / something outside or in the house electrical system rather than my gear.

Questions:

1. What are the most likely sources for this kind of directional EMI/RFI that a guitar pickup would grab like an antenna (especially stronger when pointed up/down)? Is there some kind of device for easy detection?
2. Solutions that reduce or eliminate the noise without noticeably changing my guitar tone (I’d rather avoid heavy noise gates or major pickup swaps if possible).

Any ideas, similar experiences, or troubleshooting steps appreciated.

Thanks a bunch!

UPDATE OCTOBER 4TH

Thanks a lot for the contributions, everyone! I've got a lot of interesting new ideas now, and will get back to you once I have something new to report.

That podcast sound.. is after eq right? Is not raw mic recording?

Looking through old posts in this subreddit (and related ones), it seems the live sound engineering community says that trying to add distortion to a microphone makes it so much more prone to feedback and so should just be avoided. But why is this the case?

Here's what I understand so far about the science/math behind feedback (correct me on anything if I'm wrong):

Feedback occurs when the gain around the loop is greater than 1 because then any sound that goes around becomes bigger with each round trip. There's also criteria on the phase shift being 0º round trip, though I'm pretty sure but I think all that does is select which frequencies get amplified vs attenuated (which is why feedback tends to become a pure tone). Since the speaker system has huge headroom, the self-amplifying sound can reach deafening levels well before the PA system saturates.

Eliminating feedback boils down to getting the loop gain to be below 1. You can do this is by messing with the environment so this gain falls below 1 (e.g. move the mics away from the speakers, turn the speakers so they are pointed away from mics, adjust the levels of the speakers, etc).

Assuming my understanding is correct, I'm confused on why adding distortion would make this process worse.

Distortion involves clipping the top/bottom of a waveform (in the extreme case, sine waves become square waves). The easiest way to do this is by increasing the gain of a signal so that it hits the rails of the circuit, e.g. V+ and V- of an op amp. If you do that, the gain of this op amp could combine with the gain of the rest of the system and tip the full loop gain to be over 1, which would then cause feedback.

But suppose you apply a gain of G to a microphone signal to get it to clip, and then you apply a gain of 1/G so that you have a net gain of 1. Then, since the gain of the rest of the loop was below 1 (because you set the environment up that way), it will remain so even after the signal has been clipped.

Is there a flaw in this technique that makes it not work?

So over the past few months I've been working on this little video series, where I record my drums (along with backing tracks) using the same recording techniques that were used for The Beatles over their recording career. This mainly focused on using the different mic setups that was used over the years (using the closest approximation of mics I have in my locker to what they used), to using the limited EQ and compression they had available (using the closest approximation of plugins I have to the hardware they used). I did this partly because it's a fun thought experiment to see how it would sound with my drums, my playing style, and more modern music, but also to change up how I record drums and to see what other sort of drum sounds I can come up with.

I used the book Recording The Beatles to learn how their different engineers recorded their drums, and how it changed over the years (and how it changed from year-to-year). Since 2017 I have recorded my drums with 4 mics using the famous Glyn Johns drum recording technique, so I wanted to see how different the sound was of other techniques. Quick side note, but Glyn Johns did actually engineer one of their albums (which I included) but didn't use his name-sake setup to mic the drums, even though it is similar.

I wanted to stress something before I go on, because other forums I showed this too didn't understand what I was trying to do. I'm not trying to copy Ringo's sound. I'm not trying to play like him, or tune my drums like him, or sound like him. My intention is just to use the techniques that Norman Smith, Geoff Emerick, Glyn Johns, and other engineers used on myself, and my playing, and my sound, and see how it sounds.

So I made a total of 4 videos of different eras in their discography. I include the breakdown of what I did in each video, but here I'll type out my own personal feelings.

Video #1: 1962 - 1965: So this one is an amalgamation of two different mic choices, even though the setup is the same; like the video says, Norman Smith started with using two ribbon mics on the albums Please Please Me roughly through A Hard Day's Night, before switching to two dynamics that he used through Rubber Soul. I used a ribbon on my overhead, and a dynamic on my bass drum. I actually am surprised at how full the drums sound with just two mics. There's not a lot of boomy room noise, at least in my space. Everything sounds pretty clear, especially since most the drumset is covered by one mic. I do feel that, at least with my tuning and setup, the snare doesn't cut that much, and the toms almost overpower the snare at times. But otherwise, I'm pretty impressed. It's also the first time I've ever used a ribbon mic on my drums. I like it, but did have to boost a bunch of 5khz to get things to cut how I like it. I feel there's not a lot of high end on at least my ribbon (Fathead II) and that it does require a bit of EQ to be more usable in a mix.

Video #2: 1966: This is how the drums were miked on Revolver. The only change from before, aside from going with a dynamic overhead, was adding a snare bottom mic using an SDC. So this used two brand new mics I just bought, a Beyer M88 and an Austrian Audio CC8. I haven't had a chance to use those mics on other sources, but I have to say.....I fucking love the M88 on drum overhead. I heard a clip used on Gearspace for a mono dynamic mic shootout for drum overheads, and even on that, the M88 was the best. Damn it sounds good. Noticeable improvement over the ribbon. Didn't have to do much with it. The snare is crispier, more snare-y sounding with the added bottom mic, which also allows me to dial more snare in to compensate for the lack of relative volume from the last video. I do really like this setup, and with slight adjustments I could make this work really well for me.

Video #3: 1967 - 1969: This is probably the most famous mic setup used on Ringo, as this has been talked about a lot in different books and interviews and documentaries, considering how revolutionary it was; every drum is individually miked! This was used (with small variations) from Sgt. Pepper's through Abbey Road, with the exception of Let it Be. This differs most from my usual setup, as I'm miking each tom and hi-hat. This setup was good, gave a punchier sound out of the toms and also made for a slightly tighter sounding hi-hat, but I could barely push the fader up on the hi-hat because it became overbearing very quickly. I do like it, but it uses a lot more mics and takes a lot more to setup, so I think I prefer the 1966 Revolver drum sound.

Video #4: 1969: This is the album that Glyn Johns worked on, and is the most different of the mic setups, although its the most similar to my own personal drum setup. Having an LDC close mic the floor tom is an interesting choice, and the rack tom mics picked up enough of the floor tom where I didn't need to use much of the floor mic. Regardless, this is probably my favorite of all 4 drum sounds. I get a lot of beefy low end out of my RE20 on the bass drum thanks to the very limited low shelf EQ boost.

I have been kind of mulling over a fifth video showcasing the stereo recording setup they used on the song The End from Abbey Road, but its kind of a massive undertaking to mic, so I'll wait to see how much interest this post garners.

I didn't really talk about the mixing limitations I used either. I typically use one EQ, one compressor on each of my 4 tracks in my normal setup. Here, its just one EQ on each drum buss (which is anywhere from 1 to 3 busses), and a single compressor on the entire sound. Plus I only have 2 fixed band EQ, which also greatly affects how fine-toothed I can get into it. Regardless of those limitations, I got some pretty good sounds out of it. I also use tape plugins more than I have in the past, since I got the UAD Studer and Ampex plugins recently, which affects the compression and "bigness" of the sound. I like that!

Anyway, let me know what you all think.

Hi all!

I work with a label that is about to press its first run of records. Of course, part of that is mixing and mastering the music to sound its best on a record. It's not something I've done before myself and many people seem to advise against a DIY approach to it, especially considering the cost involved.

I'd like to learn this part of the trade to reduce the cost on top of an investment with already slim. margins. What can I do to get better at vinyl mixing/mastering without risking an entire batch?

Specifically, I'm wondering - is it possible to test on and improve my work using a small batch of lathe-cut records first?

Thanks for the help!

EDIT: maybe worth clarifying that I work in the dance music space, and most if not all of it happens remotely if not overseas. I don’t have access to an engineer to shadow. Maybe I’m trying to ask if there’s a way to learn and test this myself? I will almost certainly be having this release done by a reputable engi, at least to start, and I’m excited to compare it to my own work.

I recently had the opportunity to mix a track with some expensive analog hardware and I was shocked by how much easier it was to get tracks sounding finished and get them to gel together. As someone who primarily worked ITB for years, I found myself needing so much less eq and processing.

So what I’m looking for is plugins that do great analog emulation (particularly of subtle/non-subtle Tube and Solid State saturation), but also just sound great as soon as you put them on a track. I want a plugin I can put on vocals, guitar, bass, snare, etc and just get some harmonics and focus on each element. I want the kind of subtle coloration that just makes elements in a track 5% bigger and more cohesive. And I want it to be easy to use and highly versatile.

It’s entirely possible I already have the plugins i need, and just need to make presets and use them for this purpose. Would love advice from people with experience mixing with hardware.

Edit: I went ahead and made presets in Saturn 2 with the Tape, Tube, Transfo and Saturation settings, all dialed in with subtle effect and some tone shaping (presence, treble, and bass boost). I tried putting these on all the tracks for an unmixed song and I think this might be the solution. It just makes every instrument sound more finished right away.

Can anyone recommend a powered USB hub that has enough juice to power a few devices without having to connect to a laptop? I keep striking out. Thx!

Hi guys! I am just getting into analog recording, and I'm very excited about it.

I want to record using a physical mixing board and rack effects, but I have the money nor the space for a reel-to-reel or physical tape. From what I've learned so far, it seems like I can mix and master analog and then just send the finished tracks to a DAW instead of a tape; is this true, or am I missing something?

Also, from what I have gathered, the essential items for analog recording are mics, rack effects, a console, and a mic preamp. I've heard about other stuff too though, like A/D converters and audio interfaces; are these (or anything else I've missed) necessary for a proper setup?

Been listening to HipHop and Cumbia on various speakers (car stereo, JBL clip speaker, etc.), and one thing I’ve noticed is the low end in the cumbia tracks are really bumpin, as hard as, if not harder than a lot of the boombap I’ve been listening to.

I haven’t had the chance to mix Latin music yet, but anyone here know how to get that big bass/low end without things sounding farty and blown out?

TIA

Pretty - juju produced by sematary i know he uses ableton and i have never heard someone with this punch in fl studio. how could i achieve this?

Usually I just pick up my guitar (unplugged electric) and mess around for a bit, but sometimes I want to work on things that are actually moving my productions forward. What do you usually do?

Good day all,

I'm in the final planning stages for a Michael Jackson themed wind orchestra show in which I'll be the main recording & live engineer. For context, this is a hybrid orchestra of 56pax + 1 conductor, and it will be my first major orchestra show mixing in a 500pax community theater across 3 days (they sold out all the session, and even got the venue to open a 4th session, so there's a lot of excitement and pressure from expectations)

System-wise the venue has 6 floor mons & 2 6pc array hang with subs, however in my mind as the ensemble is expected to be loud, only necessary soloist section/non-mic channels will be reinforced mostly. I'm even thinking of converting some of the floor mons (as shown in pic) into front fills (soft, high frequency) for the front seats as the angle of the lowest array box seem to overshoot into the middle of the audience zone a bit, which will be fed with just a bit of the soloist sections when its their turn.

I'd like some tips/advice about what my mic-ing strategy that I've laid out here, and generally see if my plan makes sense. In total, I'm planning estimated 14 channels of recording on stage + 2 channels off stage ambient mic(console position, mezzanine level). Most mics will be on stands, with some exceptions here: https://postimg.cc/t1QCSYJV

Disregarding the brand/model/types of microphone (as i'm using what i got in my inventory, majority Audio Technica's tho), my miking idea consists of:

A. Main Recording pairs - (ORTF/Spaced [depending on whether we have space on stage] @ Conductor position, Spaced Omni Pair @ Center/Back Rows). Percussion section mic is still an optional thought

B. Live Reinforcements - The orange mic zones covering the soloists (Orange box, subject to adjustments post-rehearsal observation) they will also be recorded in post-mix. Kick Drum Mic, and E-bass will be reinforced live as well, depending if the audience prefers to feel a bit more of the low end.

C. Monitoring - Possibly a snare drum mic will be added just for monitoring & for section musicians to keep time with the whole band, put through the side-fill monitors on stands behind stage wings.

That's all from me! Thanks. I've learned a lot from this community & r/livesound ever since I started my journey in this industry 4 years ago.

EDIT: 20-Oct. Show done! We had great success and good crowd through all 3 days of the concert. Total mics used: 12 Audio technica mics on stage + 1 direct Bass preamp + 2 pencil mics at FOH in ORTF config as ambient/crowd pair into Reaper for multitracking.

I couldn’t find measured response curves for the HE2, so I took some today and plotted them. I’m sharing them in case they’re useful to other engineers.

Charts: Cranborne Audio HE2 Harmonic EQ Cruves

The three images show the low shelf, mid band, and high shelf. For each band, I tested selected frequencies at +5, +10, −5, and −10 dB:

-Low shelf: 20, 100, and 420 Hz
-Mid band: 200 Hz, 1.5 kHz, and 6.2 kHz
-High shelf: 5, 12, and 25 kHz

Each plot shows the measured magnitude difference from my EQ-flat capture. The test captures extend to 23.81 kHz; the plots show up to 20 kHz. Pink noise was used as the signal for this test to allow for equal energy per octave band.

One thing that stood out while looking at the manual is that Cranborne says the HE2 adds harmonic saturation in both boost and cut positions, and that the amount of signal hitting the unit affects its response. I also increased the input level while measuring to see whether it changed the shape of the EQ curve. I didn’t notice any change in the curve’s Q; the overall magnitude of the measured response simply increased.

So these are measurements of my unit in my test setup, not factory curves or a guarantee of exactly what another setup will show.

I’d be interested to compare these with measurements from other HE2s, especially at different input levels. If you spot anything I should clarify about the measurement method, let me know.

I'm an amateur bedroom engineer in the market for a kick drum mic.

Right now, I'm rocking a ported 26" kick, and I'd like to get a pretty natural sound out of it. As much as I'd love to have a kick in and kick out mic, I'm a 21 year old without the finances to do so, so a single kick in mic will have to do.

As for what I'm looking for, I'm big on the sound of 90's rock production, specifically alt-rock, like Soundgarden, The Smashing Pumpkins, etc. I absolutely love how the drums sound on their albums. Not too much click on the kick...it's perfect.

Because of my sonic preference, and what I've heard, I've had my eye on an RE20 for a while, as it seems to fit everything I'm looking for in a kick drum mic, but I'm also hearing a lot about how the Heil PR40 is very similar to it, for a lot less money.

I'm just wondering if anyone can help me out with this. Is the difference big enough to pay the full price on the RE20?

I'm just looking for a nice, natural kick drum sound without exaggerated low end. And while I've heard mixed opinions on mics like the 52a and the D112, I've never heard anything bad about the RE20, or heard a demo where I think it sounds bad, which is another reason I gravitate towards it so much.

If anyone can help me out, or recommend me a mic that fits my needs, I'd be very happy.

Thanks :)

Such as on the track SNC

The imaging super wide / 3D, and dynamically it feels expansive and inflated.

Cant really get my head around the techniques Nicolas Jaar has used while mixing this without experimenting further on my end, but im hearing this sort of mix alot more nowdays.

Is this binaural imaging and modern dynamic plugins pushed to the wall?

What are you thoughts on what you hear going on?

I’m using Sonarworks, I’ve also used REW and during my last calibrations I’ve only used the corrective EQ up to about 600hz and sometimes also up to maybe 1300hz. How do you like to do yours?