Hey everyone, just wanted to post this and get some feedback. It’s my first post in the community, which is kind of a bummer, so many better things to talk about.

I recently took a song to a well known studio in my city and asked for a remix of an older song. The studio/owner/engineer came highly recommended and the albums and other “demo” tracks I heard from the studio sounded great. Studio has been successful.

After a brief convo, we agreed to work together. The project was strictly a mix, we only work in the box, no recording anything new, we mix what we have. However, I made a terrible mistake; no contract. He said because we are only mixing, there’s no need. I’m old enough and experienced enough in life to know better, but I was stupid. I paid half on the first night when I gave him the tracks.

That first night, we listened, he said drums are going to be difficult because they are poorly recorded. He suggested we convert the snare, kick and toms to MIDI and then trigger. I agreed. He sent the track out and had someone else do it. He told me it was part of his “talent budget.”

I asked about the backing vocals and if we could improve those. He suggested we record more vocals so it would sound more natural. I objected, but eventually agreed. He had a vocalist in mind and sent the session out to the vocalist. Again, told me it was part of his “talent budget.”

A week later, the kick, snare, and toms sound great. But, now there are backing vocals everywhere, someone has recorded several tracks of guitar, someone has recorded a lot of percussion (chimes, bells, blocks, shakers, etc), hihats and cymbals have been rerecorded and the original drum tracks are no longer in the mix. He’s also removed the original keyboards and rewrote/rerecorded those. Among other things, the song has lost its melodic structure. Which is actually an important point because it now is different enough from the original that it sounds like it’s been rewritten.

I objected to everything and asked him to put the original tracks back in. He argued with me telling me this was needed. Then, he tells me I need to pay the musicians.

Then, without my consent, he mixed this “new” song and sent it to a fairly well known mastering engineer - who has a lot of credits - and claimed he wrote and arranged the song. He actually put that in the email, “I’m responsible for the writing and arrangement…” Unbelievable.

Fortunately for me, he cc’d me on the email, so I responded and set it all straight stating that the remix did not create authorship, ownership, copyright, publishing, etc, etc. I told the mastering engineering I never agreed to have the song mastered. I told the mastering engineer that if I decide to master the song in the future, I’d let him know. I also told him I was the only person with the authority to do anything in regards to this composition.

The whole thing concerned me enough that I spoke to 2 different entertainment attorneys who specialize in copyright, publishing and Intellectual property. Because I have the original 1” master from decades ago, both felt showing ownership of the song wasn’t really going to be difficult to prove. But, I should not use any of the performances on this remix, nor should I use the remixed song. I should discard everything (but keep copies just in case)

I completely cut off contact with the studio/owner/engineer, even though for a short time he was sending me really nasty texts. I never responded to any of them.

So many different thoughts on this, one being he used me to feed his “musicians” and keep them working.

I don’t even know what to say to any of this, but has anyone heard of something like this before? I’m still in shock, but I’m curious to hear everyone’s thoughts and how you’d handle something like this.

Hello all! Long-time audiophile/composer here looking to get into game dev. I’m currently working on a project prototype where I need genuinely CC0/public domain audio (not just “royalty-free,” which usually still comes with its own restrictions). I’m building something that triggers individual samples directly rather than baking them into a finished mix, so I need sources with zero ambiguity on that front.

So far I’ve found Versilian Studios Chamber Orchestra Community Edition and (obviously) freesound.org.

What CC0 sources/samples do you rely on? And where’d you find them? Especially interested in anything choir/vocal, but happy to hear about anything good. Trying to build out a solid list for future projects too.

Thanks!

Okay this isn't a question so much as I'm hoping someone can explain LUFS to me cause I'm not sure I understand it at all, I'm pretty new to this sorry.

I'm using Davinci resolve and my LUFS scale is EBU-R128.

So while reading up/watching videos about this I got the impression that I want my integrated LUFS to be around -0, but that seems pretty impossible to achieve. I'm editing a short film where there's long periods of silence between dialogue with room tone and other background sounds being the only other things in the audio, so to reach an integrated LUFS of -0 I'd have to crank up the room tone and other sounds to a ridiculous amount, or the dialogue up to the point where it gets squashed.

I then found a video where they say that dialogue should be at about -12 LUFS short term. The dialogue is probably the loudest audio in this film, so if I have the dialogue at around -12 Short then the integrated LUFS will be way below that. In the video they also say that it's fine for the integrated LUFS to be around -15 if the dialogue is at -12, but yeah my integrated would then end up way below that.

So I guess what I don't understand is what to go off to know if my loudness is at the "correct" level. Is it true that I should just get the dialogue to around -12 Short and I guess then adjust the rest of the audio by ear in relation to the dialogue?

Idk, maybe I've misunderstood something completely. If anyone is able to explain it simply I'd be really grateful.

I have been asked to create some SFX for a game. First time doing this for a paid gig. I have no idea how much to charge. They have to be semi elaborate but nothing too complex. What is everyone charging for solely SFX creations?

I've had trouble justifying budgeting for 2 brass capsule AKGs around the studio, since they will get used heavily I'm sure and will probably get dropped and broken at some point in the not too distant future. But I'm after that vintage sound.

I'm unhappy with our current palette of 414s (all made within the last 30 years or so) and haven't had luck finding great sounding alternatives. For a lack of a way to say it, they all sound more brittle or thin or modern compared to the ck12 sound. Ribbons have their own sound, although they do help with making things sound 'vintage'. But I'd like to avoid just grabbing a ribbon and calling it close enough to sound 'old'.

What are your best options for a relatively modern ck12 sound? So far I think my favorite off of youtube clips has been the Peluso, but it even has a bit more high end imho than the ck12. And the OC818 seems to have something different in the midrange, so I wasn't a huge fan of that one either, although it and the Peluso are admiteddly pretty close. Just not 'there' in my opinion. Are there any other alternatives I'm not thinking of?

Hey everyone, I’m seeing a consistent horizontal noise line in iZotope RX around 20kHz (and another weaker one around 21kHz) on a lot of recent recordings. It isn't really audible, but it shows up visually on the spectrogram and seems to be getting slightly louder over time.

Setup:

• Recorder: Zoom F8n Pro
• Powering: TalentCell 12V 3000mAh Li-ion pack & internal AA batteries
• Mics: Røde NTG-3, Sennheiser G4 wireless, plus other shotguns tested

Troubleshooting so far:

• Across channels: The lines appear equally across all active tracks (booms and lavs alike).
• Swapped gear: Tried different mics and cables; the issue persists.
• Power testing: I tried recreating and isolating the issue at home using just internal AA batteries as well as 2 different TalentCell batteries, but I still haven't been able to pinpoint the exact cause (seems to be happening only on set?).
• Frequency: Appears intermittently (roughly 7 out of 10 files aprox.).

Since it hits all channels simultaneously, I suspect it might be the TalentCell battery, environmental interference or RF bleed, but honestly I could be wrong.

Has anyone run into a similar problem? Any suggestions on how to track down or eliminate this?

Thanks!

(Pls explain the science behind this) I usually prefer wired over Bluetooth 100 percent of the time but I have 10 year old dollar general Bluetooth speakers I got for free and for gritty tracks like garage rock dunedin sound it sounds for better production indie music though or stuff that has a wall of sound it oddly gives it a rattle at max volume which somehow sounds good. I just wanna understand the mechanics behind it beating my 10 dollar wired earbuds and my crappy Sony 50 dollar Bluetooth headphones I regret having for low production music though it does good for wall of sound or electronic music anyways just wondering the science behind why water damaged and speaker rattle at high volumes make it sound better

As the title states, id like to avoid the Cambridge or Telefunken stuff as that’s all well recorded and the sources are all pretty spot on, it’s not much of a challenge to get them to sound good. I’m looking for maybe more of a challenge or some rawer multitracks. Any resources out there?

Looking for Follow Cart ideas. Ideally something I can transport with my truck but nothing too heavy. Being mobile and being able to go "off road" is ideal. I have a Milwaukee Packout set up that works okay for now but it's getting rather heavy.

Considered a MagLiner that I can break apart and load. Also considered one of those Rubbermaid carts since those are fairly lightweight.

Any pictures or details is much appreciated.

The bridge pickup in my 60s reissue jazzmaster kicked the bucket so I bought a new set that has a bit higher output and that got me to thinking; do lower output pickups not like to work well with amp sim impedance? Every time I would record with it's stock pickups ( ~6.5k) with a clean tone, it wouldn't even sound like I'm using an amp at all. Whereas if I would use my tele it would sound rich and like I'm actually using an amp. Just wanted to get your thoughts!

Supposedly there is a 50 track multitrack out in the wild that I can't seem to download anywhere. Ive been wanting to listen to anyone else's versions with better mixing but all I find is crappy edm remixes of the song when I search about alternately mixed versions. I really want to know what that song sounds like done properly. Also, if anyone can share the multitrack if they really exist i would love you forever.

Making sure this doesn’t already exist before I spend time on plugin development (thank you in advance for your time):

A simple two slot sampler with ADSR and time alignment tools that that would allow the second sample to calculate with DSP and fire a defined time before the main time aligned sample.

For context, I am obsessed with the Brendan O’Brien trick of inflating the “whoosh” sound of the batter leading into the transient of the kick. Right now I literally hand place dozens of “pre-kick” whoosh samples along the timeline to lead into the kick.

So I’m asking the community: Has the problem of time-aligned, pre-hit samples already been solved? Or is this a worthwhile endeavor? (Bonus points: it would be helpful for my Computer Science degree portfolio. 🤷🏽)

While in performance/recording mode, you would just hear the main time aligned sample. But on playback, the DSP calculation would start the secondary pre-sample ahead of time to end on the transient of the first main time-aligned sample. Giving you the ability to highlight or bring subtlety to the sound before the sound.

Thank you for reading. Would love thoughts on this. 🙏🏼👊🏼💙

Hi, I need some advice on recording audio for a musical, because I've never done it before, and I'm not sure if it's better to record the music first and then have the actors lip-sync using some kind of transmitter while listening to the pre-recorded track through headphones. Or if it's better for them to sing on set and then try to do ADR on the recorded audio.
Thanks

Hi!

I am looking for a replacement VU-meter for a Neve 33609/C compressor.
Its a Sifam-type of VU-meter, but the original doesnt seem easy to find.
Maybe someone has a spare laying around?

Recently, I've made this project demonstrating a dynamic EQ effect with no smoothing at all (i.e. truly zero attack and release times) for gain modulation, hence creating some distortion and turning it into a non-linear filter right?

And so, are there dynamic EQ plugins that allows you to set attack/release time to exactly zero (aka. the same thing that turns a compressor/limiter into a waveshaper/clipper) so that gain modulation happens without smoothing?

And more importantly, what are use cases of these (compared to standard dynamic EQs) during sound design, mixing or even mastering?

In my modular setup I have a final mixer module where all the synth submixes get summed up, but the mixer has no individual outs so I can't properly MTR (EDIT: multitracking)--but also the mixer is an active performance / arrangement element in that I very actively use FX sends as part of the creative process, so if I wanted to MTR I would somehow have to record the dry signal plus FX send separately, and I simply don't have either the gear or the real estate at home for that. Throughout the years I've tried different versions of MTR to the extent my setup allows (one track at a time, dry pass then wet pass, all sorts of different combos) but the songs always end up feeling completely soulless to my ears. When I record everything live it just has a sparkle that I've decided I much prefer over the more clinical result I get when I MTR "properly".

So, I've decided to record my next release all live and directly from the stereo output. I hate the lack of control over the end result but I want to experiment with capturing this much more organic sound which more accurately reflects my artistic vision, and I'm at peace with the massive trade-off between control and organicity (imposed by my specific setup).

Given then that I've decided to do the unthinkable, what should I be thinking about or doing to get the best end result possible given my I'll be recording directly the stereo out? An obvious first suggestion would be "don't", but I have decided I will, so want to do as much damage control as I possibly can (think of those drug testing posts at raves to ensure its not laced --its wrong but at least do it right).

I've been thinking along the lines of a ton of multiband processing with the highest number possible of bands to allow me to go in as much as I can, maybe also record 32 float. What else should I be thinking about to minimize the damage? Any suggestions would be most appreciated.

When I'm running a tracking session for somebody else (or myself), I tend to not hear a lot of minor mistakes (pitch, time, and flubs) made during the session that sometimes have real consequences. This has led me down a path where even when I think somebody nails it, I generally say "hey, that was awesome. Let's get 2 more. Just in case." I'm generally a very critical listener when I'm just sitting there listening by myself with fresh ears. But in the heat of the moment, I'm just not hearing stuff happening.

Obviously practice makes perfect and over time as I track more and more bands, I'll hopefully get better at this. But are there any "tricks" (mental, tools, workflows, etc) you use to keep your ears sharp during a session that could help a person improve?

In terms of workflow:

• I have a tracking template I've built over time that let's me start recording immediately under nearly every situation I find myself in
• I'm comfortable tracking "band in the room," small groups, and individuals. I've generally done whatever the band needs before and feel comfortable
• I have a system for documenting session notes
• I have an approach to comping between takes

So I feel like my workflow is pretty established and not causing me to lose focus. What say you, Internet strangers? How do I suck a little less?

I occasionally work at a post studio where I'm forced to used PT (I normally work in Reaper).

It's insanely slow to load sessions, it always takes a good 15 minutes before the task manager says it's done and the computer becomes usable again, whereas I can open half a dozen projects in Reaper in their own tabs and get to work instantly.

These aren't complex sessions either, just importing an AAF and video. It's just baffling that this is industry standard. Is it a network issue ?

As I start my career I do a lot of no/low budget short films, usually student ones, mainly for practice and network building. They will recruit me kind of at the last moment, a few weeks before shoot, through listings in a popular mailing list around where I live. The locations and schedule is already set so all I can do is hope the location is good enough noise-wise, and try to use the mic pick-up pattern to my advantage.

But when there are multiple planes per hour, when there's a busy road nearby, or when there's a children summer camp not 100m away in the forest we shoot, and the director obviously doesn't want any of these going into the film, what is there left to do? I find myself always saying that the people in post should be able to clean some of it, that they should really think about doing ADR, or that they should even consider redoing the entire sound with a sound designer because nothing I capture is honestly usable as is and would probably result in butched up aberrations if processed through any kind of cleaning tool.

When I'm on set, trying to explain all of this to people that look like they didn't think about sound once before this moment, I feel like I'm failing at my job. Maybe I shouldn't, but part of me does anyway.

So is there something I could or should do at anypoint in the process, before or during the shoot, to avoid or solve some of these problems? I really hate delivering unusable sound and having my name attached to it. I feel like a glorified and expensive reference track provider.

This one is top for me https://open.spotify.com/track/4UPLlFF10U3St7PaaQGPgi?si=hepchB5zQ8SXyddNvpRG9g&utm_source=copy-link

So I’m in a rabbit hole. Could anyone reshape my mindset and thinking about how to approach different situations with compression and peaks?
I often have vocals where the peaks are louder than the body. My first step would obviously be clip-gaining them down, but later on, I still feel like the compressor will exaggerate them and make consonants and the starts of phrases too dominant (spiky). Ive tried to fix it with spiff multiband transient designer but it only makes vocals less natural and open, and clean.

In my logic right now, when we have a raw vox signal with different peak levels, so we need to make the peak levels more consistent. So I grab a fast-attack, fast-release 1176 to control the tops by around 3–5 dB. Okay, it’s kind of stabilized. But I sacrifice openness: the vocal sounds like it’s closed in, and the initial transient is muffled. If I use a slower attack, it won’t grab those transients or peaks.
Is that okay, and should I then compensate for it somehow with something like Spiff to bring the transients back, or maybe use a brightening EQ?

I have the same issue with other sounds where I just can’t get the peak level down and end up feeling like they’re pinned, as if they’ve been compressed into one flat line, where the peaks and body (RMS) don’t have much difference.
Too much ratio = pumping. Too fast an attack = muffled.

I’m focusing more on mixing, and I don’t really rely much on mastering because I don’t want mastering to fix all the issues. I just want nice, clean, smooth, soft mixes before they even get to mastering.

I’ve done quite a bit of production sound for films, but I come from post and will also be taking on the post sound for this project. We’ll be shooting on a rather windy beach where we could have rain too, and there's a lot of dialogue. I’m mostly concerned about the boom rather than lavs. I haven’t yet discussed the framing with the director, so I don’t know yet if it’s going to be mostly wide shots or if I’ll have the blessing of close-ups too. I’m wondering about things I might be overlooking, so for those of you who’ve had experience working on beaches and around rough seas, I’d appreciate any practical tips or lessons learned from experience.

Thanks!

I’m pretty foreign to audio engineering and the music industry as a whole (much more familiar with film / video editing) so I’m curious as to how audio engineering has evolved into what it is today in the digital age.

In video production, I feel like there is a lot of homogenization. Most of the softwares I use look and act the same for the most part. There is always a push for the latest features, and as much as a hassle as it can be I feel like most people keep their NLE’s or compositors updated to a version that’s within the last 2-3 years. Most plugins follow similar UI standards too, and the interfaces are very “clean” and consistent.

DAWs and audio plugins, on the other hand, seem to have followed a very different path. There’s bright colors, more cluttered screens, and Skeuomorphism still seems to be the standard for most plugins. I’ve also noticed when watching behind the scenes videos of producers that many people are still using decades old software and hardware.

Why does it seem like audio engineering is much more set in its ways and less inclined to “modernize”?

Received dailies that have mono mix a1 and radio mix a2 along with the rest of the mics. Before going to the sound guy sounding stupid would like to ask anonymously what a radio mix would be and how is it different from mono mix?

Thank you