Genuine question here since i am both a solo production sound mixer and a post sound designer. I have assigned on an indie horror comedy project that stated to be a series but turned to feature (we shot around may). The editor notified me around June he sending a draft and will send me update til a lock is figured out. The majority of time was wasted on me manually syncing the dialogue and sound fx (both library and foley) to pic. Til the lock is send this august and the client wanted done by july. While at the same time i was also assigned work in a short action while juggling with random call to set to work on boom op and mixing. With this much of short schedule and alot of compromise putting the feature as a priority and work on the short on different day so my rest and sleep schedule were almost non existent.
The client wasn’t satisfied what i did the sound for the feature cuz of missing note (which i followed and triple check) and now have moved on.
Now i’m just disappointed in self and try my best to finish the short today(this client was understanding on this). Please help as i love this industry and this job. Please guide me to a better sound guy

Hi Folks, back with the same old question. Right now I'm at a point where I have a solid basic understanding of both Pt and Nuendo to the degree where I can comfortably work with both, allthough I probably now know Nuendo a bit better. But I wanna really deep dive into one of them and get to a really high level. I do mainly editing and sounddesign work, but wanna do more mixing as well. I work at a studio where were doing ADR work and we work with nuendo. Like I said I can comfortably work with it, but not really sure which one is smarter to learn long term. I'm based in Europe. What do you think? 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.

edit to clarify - I’m talking about score mixing itself not anything at the edit/dub stage! Sorry if that wasn’t clear.

Curious about what others use most!

Some of my favourites -

Software:

• Manley Massive Passive (UAD) on strings bus and 2bus.
• SSL bus compressor 2 for light glue on busses.
• Shadow hills mastering compressor on mix bus for synth heavy or darker cues,
• Capitol Mastering compressor on lighter more acoustic based cues - just very light but I love the difference it makes.
• Manley Vari MU for strings warmth
• ssl subgen and/or sub filter on LF bus for easily controlled weight
• UAD Fairchild 670 on harps and nylon guitar/dulcimer
• cinematic rooms professional has basically replaced 90% of my other reverbs.

Hardware:

•I don’t have any outboard mixing gear yet but I’ve been super tempted by the Neve MBT if I was going to get anything soon!

• my most useful tool I own is my ssl uf8,uc1,uf1 setup. It has made working in Cubase in midi mode to tidy cues before exporting to jump straight to mix into protools so much quicker. Also loving working with the template set up in protools alongside my ssl controllers as I can get through the basic ground work on a cue without really looking up at the screen at all which I love and has saved me so much second guessing that I was doing when looking at plugins all the time.

•Schoeps CMC6 mk4 I love and feel I can record almost anything quickly and with really nice detail to slot in to a cue either overdubs or anything else.

Tldr:
What are your favourite tools for mixing film/tv music? Hardware/plugins, anything that takes your mixes up a notch every time! \*I appreciate the experience and expertise makes the mix not the tools but still this is just fun to hear what tools you love

Sorry for the long post! Just love learning from everyone else’s experience.

Hi all, this is my first post on this thread. I work as a First Assistant Sound Editor in Toronto, part of the DGC, been doing it for a little over 4 years now.

One of the banes of my existence is assembling an Original Sync Audio session, which is used to spot on-screen cheats (so the Dialogue Editor can match the Cheat to the mouth movement) or used to grab audio that may have been omitted by the Picture Editor (PFX, Breaths, etc) in their cut.

I of course use EdiLoad and I run either a Picture EDL (with Sound Rolls in the "Rolls" column), OR, I use a Picture AAF, both give virtually the same results. However, because these are technically pulling Timecode Metadata (Source In/Out codes) from the Camera, there's always some subframe sync inconsistencies, I think because Cameras record in Frames and Sound is recorded in Samples.

I have, however, worked on maybe a maximum of three shows ever, where the Video EDL/AAF Assembled audio which was EXACTLY in sync with the Guide Track / AAF. Everyone I've ever spoken to (APEs and ASEs alike) have no idea how to achieve it. I worked with one 1APE who had coded his own automation script to rebuild his timeline in AVID with Dailies Audio, which he would then give me an EDL export of specifically for OG Sync Assemblies, but this obviously an outlier scenario. I wouldn't expect every 1APE I work with to do this (though kind of a legend for taking the time to custom-build a script to solely make my life easier tbh)!

Anyways, thanks in advance for any pointers. People always shrug off Original Sync Assemblies "who cares if they're a bit out", but personally I find them pretty life-saving in a lot of recurring cases.

Doing a redesign for my first demo reel. Went through scenes I like from different things and tried to find a good clip without much dialogue.

The one I picked is 1:30, there’s 5 seconds where an older lady says 3 or 4 words. I am a 27 year old man and can’t imagine trying to impersonate an old woman would sound anything but silly.

The rest of the clip has so much action which did give me opportunity to do some cool stuff so I’d rather not just abandon it entirely. I also found it kinda hard to track down good scenes with 0 words in general.

Would it be best to just leave the line out since I’m trying to highlight design and not voiceover?

Hi everyone! I'm currently practicing cinematic dialogue mixing in REAPER for a medieval/epic portfolio piece.

I imagined a scene with a king delivering a speech in a massive stone hall and wanted to get opinions on my signal chain and spatialization choices. Here is what I did:

Source Chain (Voice Track): Applied a ReaEQ with a high-pass filter at 120 Hz to clear out studio proximity/mud.

Spatialization (DearVR PRO): Placed the dry voice at a distance of 6 meters for initial positioning.

Reverb Send (-15 dB): Sent the dialogue to an auxiliary track running Altiverb 7 using the "Castle de Haar Main Hall - 11.6m s-s" preset to simulate the massive scale of the hall.

Reverb EQ: Added another high-pass filter inside the reverb chain set to 160 Hz to prevent the long stone reflections from getting muddy.

Does this workflow and spacing make sense for a realistic cinematic large-hall speech, or would you adjust anything in the chain?

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Where I work we have a handful of audio engineers that each have a different idea about what stems should be, how they should be used, and how to create them.

It is my opinion that the stems should equal the mix. In order her words, if I bring the stems into a session and play them all together it should sound exactly like the full mix, and measure exactly the same on the loudness meter. The way I am doing this, prevents me from doing any master bus processing. I limit each of the sub mixes for each stem, then combine them to a record track which I then measure with the loudness meter. This also allows me to record the mix and all the stems at the same time.

Other engineers prefer to mute everything, but one stem and bounce out the mix several times. this allows them to do master processing, but the downfall is that the stems do not equal the mix.

Another engineer will bypass all of the mastering, and just bounce out each of the stems from the sub mix only, assuming that it will come back later to be mixed and mastered if something changes.

So my question to you all is: what is the industry standard? How do you all handle stems? Do you assume any changes will come back to be remixed? What are your thoughts?

Getting a new external drive to run pro tools through, and wondering if I should really be spending the extra money on a good SSD.

I feel like the answer is yes, but wanted to check here first.

I use an M4 Pro MacBook

Hi folks, I've recently found myself having to look for new full time employment, and have been wondering about finding work at an Audio Post House.

I already have experience with technical support in a studio setting (troubleshooting DAWs, consoles & analog gear) as well as some experience with my own work as a freelance sound designer

I know some folks that got full time positions working for audio branding here in NYC, but don't know how one starts to look for this kind of position... do you all have any advice on where / how to look?

I’m running an M4 pro MacBook with pro tools and two monitors, and trying to figure out the best docking station for my needs.

I basically need:

1 x thunderbolt to my interface
1 x thunderbolt to one monitor
1 x thunderbolt (or DisplayPort) to second monitor
1 x thunderbolt from my external hard drive

I guess my main question is whether the thunderbolt is crucial for the hard drive, or I can get by with just usb c? The jump in price for a 3rd or 4th downstream thunderbolt port is a lot…

Thanks a lot

I have a basic workflow, but it doesn't suit every project. Is there an easy way to create standard folders every time using a template? I want to know how to organise the Folders and Sound libraies.

What happened to the sound here? Aside from the fact that the effects in many scenes are abnormally wrong (my opinion ofc), why is it so f*\* loud? Literally, the film reaches 130dB... why is this so?

I need to stream a documentary mix with pictures to a client from Pro Tools, so we can stop and start at any point to give notes on a mix. Is anyone doing this sort of thing and can recommend a process?

I've done this during COVID via zoom and maybe that's still a good approach. But maybe there's a more suitable option now.

Obviously need to be able to communicate with client as well as sending pic + audio. I can route my mic through pro tools, and assume it'd be fine for them to unmute their mic to give feedback or 'raise a hand'

I'm working on a very important session in Pro Tools and suddenly started getting this error:

"Could not save because Assertion in './PtSess/PtSess_RunTimeShared.h', line 132."

After that, I also got:

"Pro Tools was unable to Auto Backup your session."

I'm really worried about losing my work. The session still opens and plays, but saving and auto-backup are failing.

Has anyone experienced this before?

What causes this assertion error?

Is the session corrupted?

What's the safest way to recover the session without losing my edits?

Would importing the session into a new project fix it?

Any help would be greatly appreciated.

I've had a few clients come to me recently months after completing work asking for stem delivery. They get exported along with the mix, so it's not a problem, but i can't help but feel I’m leaving money on the table.

Do you normally charge for this and if so, how do you price it?

Wave Agent is a PolyWav tool from Sound Devices that splits and combines BWFs. It is not going to work on the next version of macOS because it is intel only.

I made a small program that does a fraction of what it does, and I was wondering if there was enough demand for it to bother expanding on it and releasing it.

I've had iZotope RX 8 for years and it's always been great, didn't feel the need to upgrade. Now I'm considering getting a new computer running Tahoe so I'm trying to upgrade all my software. To my surprise, iZotope no longer offers me an option to upgrade to Rx12, it just gives me the retail price of 1,399.00 US fucking Dollars. If it were to me, I'd continue using version 8 but, of course, Tahoe OS is not longer supported. In other words, iZotope is forcing me to buy their very expensive software a second time and just throw away the 1,399 I spent just 5 years ago. This is absolutely insane.

Is this my good bye to iZotope? Probably. How's everyone doing with SpectraLayers and or Acon Digital? Any other suggestions for spectral editing?

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Welcome to the AudioPost Mine. There's going to be a lot of dirt but we hope for some gold too.

Hi everyone,

Preface, I do understand you can't mix theatrical without a theatre, this is more of a curiosity.

The consensus is to mix dialogue around about -27 lufs (guide, not a rule). Is this true for both nearfield and theatrical? If yes, is theatrical more about the dynamic range you have access to (and the other challenges with being in a large space) than where your levels should sit? Theoratically, right...? Because 60db in a cinema is 60db at home...?

Which leads me on to another question - I know we can't mix theatrical at home, but from a more logical lens, why? Theoratically 110 db at home is the same as 110db at the cinema. To that end, is it moreso technical? Does a nearfield studio space respond differently than a theratre, so it's impossible to guarentee 110db = 110db between NF and Theatrical? Maybe a dumb question but I have been thinking about it since LUFS/db/the rest are all directly correlated to each other.

To help you understand where I'm coming from, my context is mixing short films at home for nearfield, keeping in mind the viability of them being played back in a cinema while still sounding acceptable (film fests, so on). I don't care to make a "good theatrical mix", I just want to better understand the relationship taking a nearfield mix at home to a cinema and why things would or wouldn't work in my favour.

I'm in that horrible spot of knowing enough to be dangerous but being stupid enough to be wrong and useless. Any clarification is appreciated :)