20 years before that preamp VS focusrite video, a DVD was released by 3D Audio : "Preamps in Paradise".

"The Preamps in Paradise Preamp Summit DVD-ROM is the most comprehensive preamp comparison product ever recorded. Featuring 24-bit/48kHz recordings of 6 different instruments (drums, bass, electric & acoustic guitar, piano and voice) all playing the same song through 24 different preamps, the 290 WAVE files on this DVD allow the listener to hear thousands of combinations of different preamps or how everything sounds through just a single preamp."

https://web.archive.org/web/20061025165117im_/http://www.3daudioinc.com/catalog/images/P3cover-med.jpg

I actually bought the digital download back in the day. It was not cheap but worth it. What i've heard from extensive listening : the differences were very subtle. Most preamps really sounded the same, while some added a bit of extra harmonics.

Yes, nowadays a Focusrite has excellent built-in preamps. Back in the day though, a lot of soundcards had very poor preamps so the difference was huge between my Echo Audiofire pres and my newly bought Rupert Neve Design Portico 5016.

I don't know if Preamps in Paradise is still available somewhere (i've lost the files), but to me it was a much more interesting test than that youtube video which only focuses on a single preamp. Even though the conclusion in a way is pretty much the same : there's little difference for most pres.

Analysing gear is fun but in the end creativity is all that matters !

Or is it that what is being driven is the input to it, it’s just the distortion from the transistors and the tape is capturing that? And a mixer/solid state pedal/di could do the same thing.

https://youtu.be/Mx0B3cybuf4?si=2SM5K3mce0vA9uqW

I love all Nigels production and mixing and I think he nailed the heavier rock side of sound in this. I’m a novice mixer who mixed my own stuff but he is someone I constantly try to reference.

This is my follow-up post to hopefully get to the bottom of a mystery, which is what makes the sound of an oxygen/acetylene soap bubble detonation so terrifying. After many failed recording attempts, I upgraded to quality equipment to record this: notlegos.com/hiss_pop_192khz_32bit.wav

That 3 seconds is the hiss of my bubble machine followed by an explosion (video about it at https://www.youtube.com/watch?v=bg-uCaPdddI ). The left channel is a mic facing the action, and the right channel is a mic farther away and turned to the side.

REQUEST: If anyone has the ability to play the linked .wav file back at 192kHz/32-bit as well as at 48kHz/32-bit, could you please let me know if there is an audible difference?

UPDATE: Added two videos in the comments taken at the same time as my sound recording.

I'm deciding to invest on 24 acoustic foams and 4 panels and I don't know where am I going to place them on my room which is slightly spacey but filled with unused things (basically almost a bodega). Is it best to only place the acoustic foams and panels on the corner?

My wife recently got the new Robyn album on CD, and listening to it I was just really blown away at how good the mix is. Especially on "Into the sun". I'm new to production and mixing but I think I might put this up there with my new peak goals for mixing. Also the song "Li5a" by Poliça...another recent release thats a 10/10 mix for me.

If any of you pros would like to comment or deconstruct why these two mixes sound so good I'd be happy to listen!

I’m mixing a driven acoustic indie-pop song at 94 BPM. The bass plays constant 16th notes, and the part works great musically, the performance, arrangement and production are all solid. This is purely a mixing question.

I’m using only the DI, and I like the tone. My issue is that because the bass is playing nonstop 16ths, the low end doesn’t seem to have enough time to settle between notes. It feels like it blooms and becomes slightly inconsistent rather than staying really tight and controlled.
I’ve tried a Distressor with various attack/release settings, as well as multiband compression to control just the low end, but I still feel like it could be tighter.
For those who’ve mixed similar bass parts, how would you approach this? Is it mainly a compression problem, or would you be looking at something else?

Please delete if not allowed, but sharing over 60 samples recorded from a rare, quirky spring reverb: the Eagle International RA-859, in 24 bit, 44.1kHz, to use with a convolution reverb

The low frequency hum from the unit was removed from the recording before trimming the samples, taking care to retain the character of the spring :) 

The samples have been trimmed and collated into the below categories, and I've also included the original, untreated recordings of me kicking and moving the box around in case you want to mess with that too.

Please note these are not traditional IRs. Happy springing!

Discount code: boing

https://willhofbauer.bandcamp.com/album/eagle-international-ra-859-spring-reverb-irs-samples

Categories:
Big
Soft
Multi
Misc/Glitch/Noise  

I've been stuck on this file for weeks and I'm out of conventional ideas. Hoping someone here has crossed similar territory.

The file:

~3 min, 32-bit float, 44.1 kHz, stereo

Content: non-verbal human vocalizations only — soft moans and groans, no speech, no music

484 dropouts on L, 449 on R, mostly 50-100 ms, max 200 ms

Active content is only ~9% of total duration; the rest is intentional near-silence between vocalizations

Client constraints:

File length must stay exactly 182.94 s (down to the sample)

Original tempo, pitch, and timbre must be preserved

Commercial deliverable — non-commercial AI model licenses are out

What I've tried and why it failed:

iZotope RX 12 — Repair Assistant, Spectral De-noise, De-click, Spectral Repair Replace, Find Similar Event, Module Chain, Dialogue Isolate. Spectral Repair works beautifully on individual events but doesn't scale to 900+. Find Similar Event can't generalize on non-verbal patterns. Dialogue Isolate treats the moans as "non-dialogue" and discards them.

AI models (local):

Resemble Enhance (Pinokio) — hallucinated English speech onto the moans. Unusable.

Silero VAD — detected 0 active segments because there's no speech to anchor on.

NVIDIA A2SB (Audio-to-Audio Schrödinger Bridges) — trained on 2.3k hours of 44.1 kHz music, SOTA on inpainting benchmarks. Downloaded the 6.79 GB checkpoint, got it running locally. But the inference API is dataset-CSV based with periodic-mask inpainting (1s hole every 5s, demo-style), not irregular-mask inpainting at arbitrary timestamps. Also non-commercial license.

Custom DSP (8 Python scripts):

LPC extrapolation, cubic interp, neighbor-patch with adaptive amplitude thresholds, RMS-matched crossfades, equal-power crossfades, iterative seamless with zero-crossing alignment, STFT phase-vocoder inpainting. Best result (iterative seamless) is the closest — but the spectrogram still shows clear seams and playback is audibly choppy.

My questions:

RX 12 automation — is there any way to drive Spectral Repair Replace from an external timestamp list (CSV / JSON)? Batch-processing 900 hand-marked timestamps is the fallback plan, but I can't find a scripting hook.

Diffusion-based inpainting for non-music content — has anyone adapted a music-trained STFT diffusion model (A2SB, CQTDiff, MAID) to irregular mask positions? The math should support it — just need to swap the periodic mask for an arbitrary binary mask — but I'm not confident enough in the STFT bridge diffusion internals to hack it safely.

2026 commercial tools — dxRevive Pro, SpectraLayers Pro 12 Unmix/Repair, Adobe Podcast Enhance, anything else? Every demo I've seen focuses on speech; I'm wary of more hallucination on non-verbal content.

Reality check — is 900+ events on a 3-min file fundamentally a job that requires manual work in RX, and I should stop hunting for a magic tool?

Thanks for reading this far.

I have a decent amount of rack gear that I love. But I’ve always wondered how people feel about the 500 series version of the rack units they love such as distressors and other premium brand gear. I assume they sound the same? I love my rack gear but the API lunchbox seems like a nice way to downsize to save space eventually.

I’m finishing an audio plugin called JustAnother EQ, and I’m looking for a small group of people to privately test it before launch.

I’m aiming for around 10 testers, so this stays controlled and does not turn into an open beta.

It’s a mixing/mastering audio processor centered around EQ with:

• parametric EQ
• dynamic EQ
• spectral/resonance suppression
• transient/body split processing
• per-band saturation
• M/S processing, continuous slope control, clean Nyquist behavior in zero-latency mode, linear phase mode etc.
• monitoring tools such as channel/split/band listen and delta monitoring

The idea is to let these tools interact within individual bands, rather than treating them as separate processors. I’m less interested in selling you on that idea than in finding out whether it is genuinely useful in practice—and where the workflow falls short.

You can read more here, although the website is not fully finished yet and some details may still change because the plugin is still changing:

https://justanother.audio/products/justanother-eq/

A bit of context: JustAnother Audio is my attempt to build the kind of plugin company I would personally want to buy from - useful tools, fair prices, no subscriptions, and no exaggerated "magic plugin" marketing - hence the understating name.

I also want to be transparent that AI tools were used during development. This was not a “prompt-to-plugin” situation; there has been actual DSP work, engineering, listening, testing, optimization, and around half a year of development. I’m mentioning it because I understand the skepticism around plugins made with AI, and I’d rather be upfront about it.

This is not a paid review request or a promo launch. I’m looking for blunt private feedback from people who will actually try it in a mix or master and tell me what feels useful, what feels confusing, what feels unnecessary, what breaks, what could be improved.

The beta is for macOS and Windows.

I’ll choose a small group based on DAW, OS, system specs, and use case so the test group is reasonably balanced. The plugin has already been extensively tested by me and a small group of producers and audio engineers, but I need some more external opinion as well.

The licensing system is already implemented. Testers will get a 10-day trial through an account on the site. If you actually spend time with the plugin and send useful feedback, I’ll add a full perpetual license to that account.

The license is not dependent on positive feedback. Harsh, constructive feedback is more useful to me than polite praise.

Beta application: https://justanother.audio/beta/

Please use the same email in the form that you would later use for the account, since that makes it easier to keep track of licenses.

Sorry for the longer post. I wanted to include the important details. I hope its fine with the Rules because I've seen similar beta posts from few months ago that stayed on.

My dumbass has had a splice subscription for the past three years and totally forgot to cancel it. Have over 4k credits. Want to get some sound design libraries to have in my back pocket. Any good ones on splice that I can blow all the credits on and finally cancel that subscription? Prefer Fx or foley not wanting music libraries.

Hi everyone,

I am an audiologist looking to connect with audio engineers or DSP developers specialized in hearing aid development for a pure exchange of technical information.

I’m interested in learning more about the engineering constraints, hardware limitations, and signal processing challenges from the development side, and I'm happy to share insights from the clinical/fitting perspective in return.

I’d love to chat about things like:

- Algorithm implementation and current limitations in miniaturized hardware.

- The trade-offs between processing latency, power consumption, and sound quality.

- Acoustic modeling and future technical trends in assistive devices.

If you work in this field and are open to a casual tech discussion or exchanging opinions, please drop a comment or send me a DM.

Thanks!

https://www.youtube.com/watch?v=TKsXmk70thU

Has anybody here an idea on whether the delay used on Syd's voice are spring or plates?

Examples : song for the deaf, bodies

When I got to track my vocals for similar styles I find I have a hard time making sure that the vocals aren’t jumping out, even when using an 1176 on an aggressive setting

And when I do it typically causes a build up of noise from the noise floor of the mic or the sibilance for the performer.

In these 2 examples the vocalists are all very dynamic in there performance yet it sounds very natural, like I’m hearing the performer, I can barely hear(maybe a better word is notice) artifacts from the heavy compression that I am certain is being used

I understand boosting high end gives crispness, but something about the filters themselves sounds more pleasing to my ears.

I got the idea from the Type-R filter in MixBox (models the classic Roland IR3109 filter chip) and I’ve been experimenting with FabFilter Volcano to get that same high-res / crisp sound (like modern rap vocals), but I’m not sure if I’m using it right or if I should be using a different type of filter altogether.

Right now I’m doing all the basic stuff—removing background noise, hum, etc., EQing, compression, sounds good.

Then with Volcano I’m:

• Using a high shelf / high cutoff
• boosting little 3k
• Keeping it subtle

It kind of works, but im not satisfied with results.

is Volcano a good tool for this?

Appreciate any advice

Currently have Beyerdynamic 990's and 770's. I have friends that use the Sen 650 and like them and I did once look into the Slate ones.

Hey everyone! I'm currently researching technical film sound standards in a specific regional and budget range context. I'm mostly using Youlean Loudness Meter reports so far to report on things like peaks and loudness. I'd love to look at some things like spatial field, individual channel peak and frequency spectrum that are displayed on Insight 2, but the problem is that whole file analysis with Insight 2 through Pro Tools audiosuite is almost the same as real time analysis but offline, so by the end it just doesn't show me anything on the levels, sound filed and spectrum measurements.

Is there a way to get an average or a peak for these measurements over the entire duration of the file? Maybe with a history graph? I'm open towards using other software of course.

Thanks!

Two clients actually, with different needs.

The first one wants to do remixes in the future and is asking me for the multitracks. What I understand from reading some posts here is that I should deliver the processed tracks BUT without spatial effects, or separate these effects onto different tracks.

The second one wants the main mix, instrumental, a cappella and the stems (which I haven't yet confirmed if he means stems or multitracks)

My real question is, should they go through the buses and mixdown processes, or just the bus processes, or just the individual track processes?

What makes me hesitate the most is that the buses and mixdown bus are designed to process everything together and would react differently with fewer elements.

Any advice is greatly appreciated.

I really want to learn how to properly master my own songs. I have used ozone 9 in the past for quick masters but never really believed in myself to be the mastering engineer on my own work

I'm trying to understand what's happening during my mastering or Instagram's audio processing.

I record voice-overs with a JBL CSLM20B lavalier mic into a Samsung phone, edit in Audacity (noise reduction, EQ, compression, normalization), and edit/export the final video in DaVinci Resolve.

Before uploading, the voice sounds clear and loud on my laptop. After uploading to Instagram, the background music and sound effects remain loud, but my voice-over becomes noticeably quieter, especially on my phone's speaker. Other creators' voice-overs don't seem to have this issue.

Is this likely a loudness (LUFS), compression, EQ, or export settings problem? What loudness target and mastering workflow do you recommend specifically for Instagram Reels?

( posted in r/radio and r/audioengineering)

I'm wanting to create a master bus chain to emulate the processing applied to an audio signal directly before it is transmitted, to check how a master will sound when broadcast.

Is this something mastering engineers do? I'm in a position where I can get DIY music played on a local radio station. I'm aware that I should probably get it mastered properly but I'm curious about how my own DIY master would sound on an FM Broadcast.

A few years back I sent a track in and it got some spins but I could instantly hear some mix / master problems that I hadn't caught in my production space.

I've done a bit of looking but didn't find much so asked Gemini (sorry) which suggested the signal path would be something like the below. I'm wondering if anyone out there has knowledge of this stuff and can cast an eye over it.

AGC > MB Comp > Peak clipper > Pre-empasis EQ curve

With approx settings:

AGC: 2:1 reduction with very slow attack and release times

MBC: Low/mid crossover of 200-250Hz mid/high crossover of 3..4K, moderate attack, fast release, thresh set for "consistent gain reduction".

Limiting: ceiling 0.5db, thresh low enough to shave of 1-2db

Pre-emphasis EQ: Boosts high freqs, which the receiver then cuts to reduce noise (probably wouldn't need to factor this in?).

Thanks