I've been going back and forth on this lately, especially with multi-mic drum recordings. It's pretty easy to look at the waveforms, move everything into alignment, and assume you've made the drums tighter. But sometimes the result sounds smaller or loses some of the character that made the kit sound good in the first place. I've been experimenting more with phase correction instead of just automatically time aligning every transient. The difference can be pretty noticeable on the kick and snare, particularly when you're balancing close mics with overheads and rooms.

For those of you mixing drums regularly, what's your approach? Do you always time align? Do you manually adjust polarity and phase, use a plugin, or mostly trust your ears and leave the timing alone? I'm particularly interested in workflows that improve low end punch without making the whole kit sound unnatural.

Hello everyone!

I have enough money set aside for essentially one solid online course, and I want to find a course that will be most worth my while.

For context, I work full time teaching basic music production at both the primary school and university level. I live in the LA area and love attending synth expos and meetups, etc. I also produce my own music and hope to record my first album within the next year or two.

I really just want to take my knowledge to the next level. The main courses I am considering are mixer brain from Jeff Ellis, Greazy Wil’s courses (particularly the recording one) and of course Think Like a Synth from Anthony Marinelli.

My question is, would any of these (or other courses be worth the $400-500 they cost, or would I be better of learning for free online and putting the money toward something else?

Thanks!

We are talking up to 8 hours of sustained 130-140F. Obviously acoustic guitars need to go and probably my synths too. I'm still unsure about the GIK acoustic treatment. I'm very unsure about my E8XT monitors and Kali IN-5's.

Is there a program or plugin that can analyze the pitch of multiple samples at once?

I have a ton of one shot kick and snare samples, and it would take forever to check each one separately.

The way I do it now is to load a sample(s) to an audio track, then translate it in Melodyne. And then I have to go thru each one individually and rename the file....

Thanks.

Hey everyone,

I'm a 21-year-old software engineering and computer networking student based in Europe, and I want to break into the live sound industry. I have a massive passion for acoustics, physical properties of sound, mixing, and system engineering. I also have a tiny bit of practical experience from organizing a one day music festival, underground raves, playing in amateur bands live and music production in Reaper as a hobby.

I have a direct connection to the owner of one of the major regional rental companies (they do big arena shows, corporate events, and festivals). I’m planning to sit down with him soon to ask for a foot in the door.

I want to be brutally honest with myself: I am 100% aware that having an IT background doesn't make me an engineer on the field. I suppose that for the first year or two, my job will be pushing heavy flight cases, wrapping muddy cables at 4 AM, and doing the dirty work which I am completely fine with. I understand that college teaches you practically nothing and I have a strong will to learn. I am brutally afraid of monotonous office IT jobs.

However, my long-term goal is to leverage my networking and software knowledge to become a System Tech / Audio Network Engineer.

Since all of the experience I have gathered during college is basic networking, systems administration and coding (basic as in elementary network setups, wiring, OSPF and VLAN configuration, DNS, ...) I suppose that the learning path will be rough.
I am looking to narrow down to a structured and well tested learning path.

I’m currently planning to get my Audinate Dante Level 1 & 2 which will be my first ever certs.

My questions for the experienced folks here:

1. What does the learning path look like for someone transitioning from pure IT/Networking into System Tech roles?
2. What are the absolute "must-know" concepts in acoustics and audio electricity that my IT college didn't teach me?
3. Which software should I start messing around with at home? (e.g., Soundvision, ArrayCalc, SMAART, SysTune, ...?)
4. How do I balance my IT brain with the reality of the field so I don't end up the "boss's little boy"?
5. This one I feel is the most important for me. Should I first brush up on specific details and knowledge needed for the role before actually contacting the boss or should I just do it and potentially learn there?

Any advice, book recommendations, YouTube channels, or harsh reality checks are highly appreciated. Thanks!

Hi everyone new to post production for audiobooks. Just picked up a gig where I master the audio for these long form books . What plugins are you guys using to hit the levels on the long form audio format? The levels they want me to hit are -16lufs and -2 true peaks. Is there a way to batch this for the files? At the moment I have 50 files and they are all supposed to hit this range. Am I overthinking it? What is the easiest way to do this? I am on pro tools of that makes a difference. TIA for any help.

I normally mix and listen on headphones (Sony MDR) because I have pretty severe high frequency hearing loss (roll-off around 1kHz coming to a complete cut-off point by 4kHz. Though lately, I've pulled out my old monitors (JBL 305's) and started experimenting with those again because I really like the stereo imaging of speakers.

My point of frustration is this: I really like not wearing headphones, and I really like how stereo sounds on speakers, but the problem is that it seems like whenever I'm trying to mix (or even just listen to my collection) on these monitors, I can't hear nearly as much detail as I can on the MDR's. The sound is much darker, and transients don't poke through very much.

That said, I fully acknowledge that my room is untreated. It's probably a good 16x20x14 bedroom and the floor is concrete. There's tons of stuff along every wall, so I'd like to think that helps a little bit, but probably not much.

So I guess I'm posting to ask, what do y'all think is the issue here? Is my hearing probably just too bad for me to use monitors effectively? Or do these problems sound like typical untreated room symptoms? Do you think I just need better/different monitors? Turn the volume up more? Skill issue? Would love any suggestions from more experienced people

EDIT: In case it's relevant, I'll just mention that I have noticed that my listening experience is much better when I EQ up the mids in iTunes' equalizer. It's not perfect, but it does seem to sound remarkably closer to how I'm used to things sounding on my headphones. I'm not sure if this is a clue that it may just be my ears

I recently was able to upgrade my set up and got some outboard gear. I got to test it out with one of my clients but I wasn’t getting signal in pro tools, here’s the set up:

Condenser mic > neve pre amp (mic level) with 48v > distressor compressor > Apollo x4 (I thought the pre amp brought up to line level) > MacBook

So basically I wouldn’t get signal unless I put the apollo on mic input, but I thought the preamp brought the signal to line level. What am I misunderstanding. I appreciate any help! Also first time posting on Reddit. Prepared to get roasted lol also now I’m scared I fucked up the mic. Still worked though, I kept the level pretty low to avoid any pops n stuff.

Free tape machine plugin, released today, haven't tried it but excited to give it a go.

https://www.patreon.com/analogobsession/posts/themachine-173345538?utm_campaign=patron_engagement&utm_source=post_link&post_id=173345538&utm_id=7553d3e5-e1b2-4866-8e8b-ae594e42b481&utm_medium=email

Hey everyone!

I've recently been getting more serious about audio engineering, specifically mixing vocals, and I'm looking to start investing in professional plugins that I can use long-term.

I'm currently deciding between the FabFilter Mixing Bundle and the Horizon Waves bundle I've been looking at.

Since I'm a university student, I can potentially get FabFilter's Mixing Bundle for around $365 with their 50% educational discount, compared to the $299 Waves bundle I was considering. However, I'm sure you're aware of the massive amount of plug-ins in the Waves Bundle (which I assume I wont use)

I'm particularly interested in mixing atmospheric R&B and darker, more experimental music, similar to The Weeknd's earlier work. I really enjoy spacious vocals, heavy reverb, delays, and creative vocal processing.

One of my biggest concerns with Waves is their Waves Update Plan (WUP). I've been reading a lot of criticism about having to pay for updates after the initial coverage expires, especially when upgrading operating systems or switching computers. Since I want to build a collection of plugins that I actually own and can reliably use for years, I'm wondering how much of an issue WUP really is in practice.

My current plan is to purchase the FabFilter Mixing Bundle as my main set of mixing tools, then gradually pick up individual Waves plugins during their promotions rather than buying an entire bundle filled with plugins I might never use.

However, I'd love to hear from people who have experience with both companies.

• Is FabFilter noticeably better in terms of audio quality, or is the main difference the interface, workflow, and flexibility?
• How has your experience been with Waves' Update Plan (WUP)? Has it actually caused you problems or additional expenses over the years?
• Would you personally choose seven FabFilter plugins for $365 over a larger Waves bundle for $299?
• Are there any specific advantages to Waves that you feel FabFilter doesn't offer?

Ultimately, I'm trying to build a reliable plugin collection I can use for years on both Windows and Mac, without constantly having to pay for upgrades or plugins I don't need.

I'd appreciate any advice, especially from people who have worked extensively with both companies. Thanks!

My audio interface does that, a 2 db headphone output channel missmatch.

Woud you care or just ignore it because it's too little? The inputs don't have any missmatch

Edit: Turns out the entire audio interface suffers from this channel missmatch. I’ve just returned the audio interface to Thomann. The engineers there told me that it was a manufacturing issue with my unit specifically. Once a new unit arrives, I’ll purchase it again to see if it’s fine or not

I'm mixing an afro track (wait - hear me out) and I've been mixing with the subs off on this song because the magic is in the midrange and all that however I turned on the sub and suddenly the kick was just diming the subwoofer cone so I backed off on some previous EQ moves I made in the subs on the kick but then I'm kind of staring at the meters (I swear I'm not a midwit and I'm going somewhere with this) and I'm thinking, the snare 'looks' louder and the kick still sounds anemic... I figured let me throw on some compression (ofcourse bearing in mind that compression takes some low end away) and suddenly it's like... I'm realising without a sub/accurate low end represntation you are not hearing the compression of a low kick.

Yes, you can compare the isolated subfrequencies and use references as a safety net however every mix is done in a different room on different speakers and then likely mastered on a different system. So now you're what, trying to achieve a mastered sounding mix? Seems risky if done without forward thinking because suddenly you might be in a situation where there's no room for the mastering engineer to actually utilise dynamics. This is where things like faux mastering and master bus processing can help because then when your mix that's going into compression works against a master, you know when you send the uncompressed version to a mastering engineer, they'll actually have room to work. Ofcourse mixing into that also can have it's problems if again you're not forward thinking. That's two seperate philosophies though, mixing for mastering and leaving nothing for mastering.

Nonetheless, I digress. Low end is what mastering engineers complain about the most besides overall dynamic range. I got REW and a measurement mic and everything is calibrated to give me a flat response. Room is lightly treated. And I'm just thinking like... hearing low end is so much more critical than we give it credit. I'm one of those people who can't build a mix on headphones but don't mind detail work and checks on them so keep that in mind. I'd argue a sub is not something you should just mix into necessarily either. I'm of the old philosophy of starting the mix on the small speakers, spending majority of time on near/mid fields and then moving up to the far field monitors until you find a good medium that works. I also check my mixes on IEMs, open back headphones, bluetooth speakers and mono PA speakers. Getting that low end wrong though bites you in the mastering stage because now it needs to be fixed and that means you're giving up some of the intentionality which is the point of a mix. I'm starting to think this is why people's mixes fail in the car - because the car has subs! - not to mention masters are compressed or limited for loudness. If you're clever you can definitely mix in the car to dial in the low end but that's a topic for another day.

I think I've been taking my sub for granted because low end translation stopped being a problem when I got subs. The meters didn't necessarily matter - I like loud snares - however they helped me confirm there was a problem. Without the sub it'd be a game of listen in the car or other capable system, make notes, try something that works and call it a day instead of the ideal of being able to actually dial in something that sounds great. Worse yet if you never check and the balance of your mix sounds fine meanwhile the mastering engineer sets off his car alarm when the beat drops after the intro. Obviously this works for me. I mix a bit of everything. I know it certainly doesn't matter as much in some genres. I know some people can't mix with subs or prefer mixing on headphones. Anyone else very pro sub though? Thoughts and critiques welcome.

EDIT: Since some people seem to think I write for Sound On Sound, I put some paragraphs in there. Just ask nicely. Some of y'all are rude as heck x_x

Hello, writing here on a very rainy lazy Sunday. Before I burn more hours on this I want to know if I'm even able to achieve what I would like;

I have a model 24 connected to a Lenovo X260 and that's running the latest Reaper version inside Linux Mint. Essentially, my instruments sound great when monitored from their Tascam channel, the return audio from Reaper into channels 1 and 2 is, well, weak, flat, very meh. I don't think it's possible but can the EQ in on the Tascam record into Reaper or do I need to EQ in Reaper too? Or am I missing a trick?

And why? What are some other nice bits of functionality that it has as well? I use Pro Tools.

Hey everyone! I'm working a new track of mine, but I just could not get it to sound this big.

The reference song

I have no idea what I'm missing, like what type of reverb should I use? where should I use them.

The audio in Evil Dead 2 has some very unique distortion. What causes it? Is it a low record speed or could it be some sort of flanging? You can hear the distortion at 0:54 in the linked video when the foggy titles appear. I'd like to have a similar sound in my film projects, so any info would be helpful. Thank you.

Let’s discuss speaker placement and the equilateral triangle and all the variables that are not discussed, and actually how important it is.
Yes, we know about the equilateral triangle and what is suggested for monitor placement, but what isn’t necessarily discussed is where this measurement is measured. Also where is the point of the triangle meeting on the listener, behind the head, the middle of the head, at the ears? Various sources say conflicting things, so what should we follow?

I have Dynaudio BM15a’s. They’re fairly large for a near field. The BM15a’s manual gives a broad listening distance of 1-3 meters. If you’re unfamiliar, the BM15a’s also have an offset tweeter, sort of like an NS10.

When researching speaker placement and the equilateral triangle, there’s usually a graphic example showing the measurement between the two speakers and the listeners head, with the measurements starting from a drawing of two single speakers that look like drivers. However, other sources say the measurement should be coming from the distance between the two tweeters. While other sources say the measurement should be coming from the distance between the two acoustic axis points. This sounds tedious, but for me it is. Having offset tweeters, they’re in a different place than where the acoustic axis point is and also where the driver is, so finding the equilateral distance is a little more difficult for me.
Also the point: the point behind the head. Like said above, different sources say different things. The point where the triangle meets the listener is either the ears, inside the head, or just behind the head—so which is it? Varying sources also talk about how the tweeters should be pointed at your ears, while others say the tweeters should be going just past your ears.

So here’s my concrete questions:

1. Where is the measurement between the two speakers starting? The tweeter? The acoustic axis point? Or the drivers?

2. Where is the point of the triangle meeting the listener? Is your head the point? Or is your head living inside the triangle? And if so, how far behind the head is the point?

3. When toe-ing in the speakers, are you pointing tweeters at ears directly or are they going just beyond?

4. (Bonus question) does any of this actually matter?

I’m looking into converting a home office (technically a bedroom) into a home music room. While I will be doing some rough recording (somewhere in quality and enthusiasm between recording into a phone‘s voice memos and something semi-professional), the main purpose will be to make music being provided sound best while listening live.

I have voice lessons in a small room that has rather minimal acoustic treatment: some burlap coffee bags with foam inside on the wall across from me (and maybe behind me). Nothing on the side walls other than a mirror. Instructor sitting at a piano between me and the back wall. Nothing fancy, but I swear my acoustic guitar sounds so much better than anywhere else. It’s so full and rich sounding. I don’t know how to describe how the room sounds generally other than like muted and soft. It’s hard to really judge how much better/different my voice sounds. But the room sounds less echoey. For what it’s worth, I think my instructor said the interiors of the walls also had some soundproofing elements when constructed. The door is very heavy.

So, my primary goal is to get things sounding closer to that room for live listening. Getting the room to sound okay for recording is a secondary consideration. For what it’s worth, I recorded myself into a mixer into an iPad through a condenser mic in my small walk-in closet facing racks of clothes, and I though the audio quality was more than adequate if not great. I record mostly as a learning tool for myself. It doesn’t need to sound professional, just a relatively accurate representation of my voice.

The room is about 10’x10’ with an 8’ ceiling. I took some. I can’t seem to add pictures of the room. I can try to describe the layout if that helps. I’ve looked into acoustic treatment articles and most focus on panels relative to the monitors/speakers as part of a recording studio. I’m not sure how to translate that to my situation. And when it comes to recording/mixing, I’ll probably mostly use headphones when listening back.

So, any advice on how to acoustically treat the room so acoustic instruments sound natural and rich would be wonderful.

Apologies if this isn’t a good place to post this question, but was eyeing the FabFilter Pro Q 4 and awaiting Black Friday, when I came across it [on Muziker](https://www.muziker.com/fabfilter-pro-q-4) for $135. Seeing as it’s normally $200 and likely would be 25% off on Black Friday, this deal seems a bit strange to be so low.

Is this legit or should I be concerned?

Hey everybody, I'm OP and I've been into audio engineering as a hobbyist since I was 11, so about 16 years.

I'm told I'm really good at it and had about a year of professional experience before deciding pro work wasn't for me.

My two friends are avid music listeners, these guys are the fans every artist wants.

The opinion: Music DOES have a right and wrong, it's just that sometimes the wrong becomes the right but only if it's done right.

The context: The blown speaker, clipping 808, clipping distorted vocal sound in HIP-HOP was not intentional, it only became a trend when avid listeners didn't care that it was mixed like shit. THEN professionals applied the right techniques to get a quality mix using the inspiration. Only then did it become right but it's still technically wrong, just with a right way to do it.

Their Opinion: There is no right or wrong in art period due to the fact that it's art.

Things that make sense in context of your mixes but might not be the default way of doing things.

i'll start:

i prefer expanders over gates because they dont struggle as much with the quieter drum hits.

i frequently reverb out the snare bottom mic (100% wet). it gives a good high end sizzle.

I do more clip gain than volume automation.

I use more busses than necessary.

Hi, I recently finished welding up my own stainless steel waterphone, an instrument commonly used for horror movie foley.
I have a clip on contact mic for it but while it does pick up on the sustain of the instrument it doesn’t pick up the initial strikes on its tines as well.
What is the best method and hardware to mic this with?

i’ve been trying to replicate this exact effect https://youtube.com/shorts/sf0wOue0yEo?si=vBwdoHY_Ya8TDvlV ..I asked a couple of freelance engineers to do it and all they did was slap reverb on the voice and call it a day 😭. It sounds like mud. I even tried throwing a random cathedral reverb and adding a basic delay, but it just doesn't feel right. ps Im not an audio engineer, i just make videos and edit both video and audio on davinci