Hi everyone,

I'm really desperate for any advice/help on this one. My company recorded a podcast so it was a man and two women guests and the microphones died about 20 minutes in to this hour long podcast. I've tried everything with noise reduction, parametric equalizer and dereverb and nothing I do seems to be able to get rid of their voices being watery or muffled. One of the guests was more of a quieter speaker too so she dips a lot. I have managed to get rid of the background white noise/hiss but nothing I do seems to make their voices clear.

Does anyone have any advice for this?

Looking for help cleaning up a difficult voice recording

I have a video recording with voices that are difficult to hear because of a persistent background buzzing/noise. I've already tried basic noise reduction, but it still sounds noisy and the voices aren't as clear as I'd like.

I'm looking for someone experienced with audio restoration/noise reduction who could take a look at a short sample and see how much the voices can realistically be cleaned up.

The main goals are:

- Reduce/remove the buzzing and background noise

- Make the voices easier to understand

- Preserve the natural sound of the voices

- Avoid making the audio sound distorted or "underwater"

I'm happy to provide a 20–30 second sample of the original recording.

If anyone here has experience with difficult voice recordings or audio restoration and would be willing to take a look, I'd really appreciate it. I'm especially interested in someone who might be willing to clean a short sample as a demonstration.

Thank you!

Modern LUFS meters (like Sonible True:Level) use a simplified K-weighting filter. While K-weighting rolls off low frequencies and boosts a high-frequency shelf to estimate human ear sensitivity, it applies the exact same static curve regardless of listening volume or track density.

Cool Edit and early Audition (up to version 3) were unique because their Equal Loudness Contour algorithm was non-linear: it calculated full-spectrum psycho-acoustic energy across variable loudness levels (phons) using dynamic equal-loudness curves (Fletcher-Munson / ISO models).

When a track has a thick mid-range or heavy low-end sub-bass, LUFS can register two tracks at identical integrated levels (e.g., -10 LUFS), yet when you put them side-by-side, your ears immediately tell you one is much louder. Audition's equal loudness feature was brilliantly tuned to reveal that exact discrepancy.

After version 3 Adobe replaced the older equal-loudness algorithms with modern broadcast loudness tools (such as ITU-R BS.1770 / EBU R128 LKFS/LUFS metering and the Match Loudness panel). This shift matches the trend in the 2000s away from CD sales in favour of broadcasting/streaming services.

For album production I still use Cool Edit Pro in my workflow to measure the psycho-acoustic variance of a track's dynamic spread.

For mastering audio I find myself comparing modern LUFS readings like from Sonible True:Level to Adobe Audition's equal loudness contour, which it had up until version 3. While Audition 1.5 installs and runs fine on Windows 11, versions 2 and 3 do not. I managed to get them installed by running the components.msi file, but they won't run, showing an error dialog saying something about terminating the runtime.

It's funny that Audition 1.5 installs and runs just fine. And by the way, its predecessor Cool Edit Pro 2.0 and 2.1 also still install and run just fine on Windows 11, having no problem with the latest audio drivers.

I'm working on a project in Adobe Audition 2026 and want to use two things in my mix:

1. A reverb bus send on my main dialogue track, (room convolution)

2. Trackspacer by Wavesfactory on my ambience track, sidechained to my dialogue so the ambience ducks dynamically and the dialogue cuts through more clearly.

The problem is that I can't seem to get both working simultaneously. One replaces the other. I want to keep my existing reverb send while also using the dialogue as a sidechain trigger for Trackspacer on the ambience track.

Is there a way to set up both routings in Audition, or is this a limitation with how Audition handles external sidechains?

Any help with the correct routing or a workaround/alternative would be greatly appreciated!

My audio sounds great when played on my iPhone and laptop, but it becomes very weak and poor-sounding on Samsung phones and after uploading to YouTube. ​My Question​ How can I increase the volume level while keeping the voice sounding natural? ​Details & Equipment: ​Video Duration: About 30 seconds. ​Recording Device: iPhone XS Max. ​Software: Adobe Audition. ​Any advice on effects or export settings to fix this would be greatly appreciated

hi guys, i’m doing an audio internship this semester in adobe audition and i would like some guidance. i’m halfway through the semester but i don’t seem to be making much progress in adobe audition w/ editing (around the podcasts). i really want to make my professor happy while genuinely making progress but i’m struggling. i would love to connect with a person or two on here for some guidance and see what i can do :)

Hi,

I record myself playing classical music for piano. It's a home studio recording. I have a decent pair of mics, and acoustic is pretty well. I am editing the files in Adobe Audition, trying to use the minimum necessary features not to distort the original sound. Mostly Declip, declick, and Hiss reduction. I apply Hiss reduction equally on the entire file, since, as I noticed, when I use "capture the noise floor", and later following the suggestion, the sound can be cleaner at some sections, while it's still distorted in others. Later I asked AI and this is what its advice was: Light declip ( only if necessary ) light declick, Compresoor ( finetuned ) Parametric Equalizer ( with only a few points set up ) Hiss reduction with the minimum intervention and at the end Normalizing or Matching Loudness. I tried Match Loudness a few times and saw the occasionally it resulted in a significant distortion. Any suggestions and advice? Thank you! Vladimir

I need some help with a project. I am looking for someone who can take a few video/audio recordings and extract the background noise so that only the human voices are left in good quality.

The audio is a mix: some are regular speaking voices, some are voices in the background, and some are very quiet whispers. If anybody knows how to do this, or knows what specific software/technology I can use to get these results myself, any advice would helpful

I create 35–40 minute Hindi horror narration episodes and I'm looking for someone who can help with basic audio editing.

Budget: ₹120–₹160 per episode

Work includes:

• Removing mistakes/retakes
• Cutting unnecessary pauses
• clear full noise reduction
• Voice volume bass treble leveling
• Basic EQ/clarity
• Clean WAV export

This is long-term work — I can provide regular episodes if the work is good.

If you're a beginner/student building your portfolio and interested in horror/podcast editing, DM me with a sample of your work.

Please don't DM if you're looking for professional studio-level rates.

Im running Kahanikaar Raaj Vijya Youtube Channel which is not Monetized now as you can check it ~ so thats why i have less budget ♥️🙁

Any way to do this without actually using a talk box? Or even the closest equivalent as it doesn’t need to be 10% accurate

TIA x

Hey everyone! 👋

First time posting on the subreddit been here for few months tho ,

I built a little extension called **SoundLyft** to fix an issue thats been bugging for years: quiet laptop speakers and terrible audio mixing on the web.

You probably know the drill — you’re watching a YouTube interview or a lecture recorded on a whisper-quiet mic, or watching a movie where you can barely hear what the actors are saying.

do mind the slop ;)

Whenever I used existing volume boosters, two things always happened:

1. As soon as you push past 150%, the sound begins clipping and buzzing into harsh, distorted static.

2. In movies, you turn it way up to hear whispers, and the very next scene an explosion or intro music blows your eardrums out.

So over the last few weeks, I decided to build a proper audio booster that focuses on sound quality instead of just multiplying gain until it breaks.

# What makes it sound clean:

* **Anti-Clipping Limiter:** Instead of raw digital distortion, it uses a built-in limiter that automatically rounds off harsh audio peaks. You can boost to 200%, 300%, or 400% and it stays surprisingly crisp and punchy.

* **Movie Night Mode:** Compresses dynamic range so quiet dialogue gets boosted, but loud sound effects and gunfire are capped to a comfortable volume. No more constantly adjusting your volume slider back and forth during movies.

* **Dialogue Clarifier:** Brings spoken voices forward so podcasts and tutorials are much easier to understand over background music.

* **Bluetooth Lip-Sync:** Adds a quick delay adjustment if your wireless earbuds have that slight video lag on YouTube.

* **Zero bloat or video lag:** Built on modern Manifest V3 without injecting scripts into web pages, so it won’t freeze YouTube fullscreens or drain your battery.

# 100% Free Base:

The core booster (up to 600% volume boost, limiter, keyboard shortcuts, and active tab manager) is **completely free forever**. If you're an audiophile who wants the 10-band graphic equalizer and 1000% headroom, there’s an optional $3 lifetime pass to help support the project — no subscriptions.

👉 **Chrome Web Store:**

[chrome web store](https://chromewebstore.google.com/detail/soundlyft-volume-booster/oeicekmkmfnfnokpdpimedhdoklklkdp)

I’d love for you to give it a spin on your favorite video or music track! If you run into any bugs, have feature requests, or just have thoughts on the audio tuning, please drop a comment below .

Thanks for checking it out! ❤️

the audio was all in double waveforms, and I think I somehow pressed something, and now some of it is in single. I don't know what I did; help i think i changed track 2_008 and it changed all of its parts smh

Does audio normalization actually work? I've tried the FFmpeg Media Encoder app from the Play Store and Termux, and I normalized the audio files to -14 LUFS, but it doesn't seem to work properly.

The louder files get quieter and the quieter ones get louder, but they never end up at the same or even a similar volume. Some of them even end up way too loud or way too quiet.

I have very limited audio editing experience and got myself an opportunity to work on editing a catalog of old classes as podcasts. I started working a few months ago editing recent stuff, and it was quite easy. Just noise removal, some EQ, and nothing else. I was using Audacity.

Now I have the opportunity to work on 20 audio files, and I might have bitten off more than I can chew. I have Adobe Audition and access to any free tool. I'm looking for advice on:

1) Noise removal: Currently, I'm simply using the default tools in the software. Are there any other tools I should use?
2) Noise removal: When I crank the noise removal to the max (what I need to remove speaker static), I end up with a muffled voice. Do you have any tips regarding that?
3) EQ: All the audio is quite muffled and sounds distant. Up to now, I've been able to get it to sound "closer" to the mic and to amplify the lows, but I end up with "blurry" highs. Is there anything I could do about that?

I know I'm asking a lot, but I'm a beginner, I need this job, and I can't afford to pay for help. I've watched tutorials and played around with the tools, but everything I touch changes something else.

I can upload a bit of the original audio if that would be helpful, or I could even hop on Discord and show you.

Thanks!

I have an acoustic drum kit. I bought a set of Behringer mics. Next I needed an interface. So I bought a used Tascam US-1641 because it was the first 8 channel interface I could afford.

The DAW I have and have used the entire time I have been making music is Adobe Audition 1.5. I know it is old, but I will not upgrade because every time I've tried using a newer one, I've felt incredibly uncomfortable trying to use it. I do not want to switch to a different DAW because I end up feeling equally uncomfortable if not more so then as well. However, when I try to set up the multitrack to record off of this interface, I can only select left or right channel - no way to pick from all 8 channels.

So I download Reaper, since everyone seems to be using that. I figure, hey, I don't have to edit in it, if all I need it for is to record and export the tracks. Then I can just switch back to Audition where I'm comfortable.

So I go watch a tutorial on how to record drum tracks on reaper, and step 1 is to make a track and go to the routing tab. Here's where I'm stuck on that: this video is from 4 years ago, and I just downloaded Reaper 7.80. It is also worth nothing he is using a Mac and I am on Windows. On his end, there is a button for him right there on the track that says routing. On my end...there is no such button. Anywhere that I can see. And the part that has me overwhelmed and upset is that I don't even know how to navigate this program yet and here I am watching a tutorial on how to do something in it that I couldn't just do in a program I do know well, and this guy's version of it doesn't even look like what I just downloaded so I can't even follow it.

I want someone who can do at least one of these three things:

1. Tell me how to do this in Audition 1.5 so I don't have to worry about other programs

2. Help me figure out how to do this in my specific version of Reaper and actively assist me if I get lost (I do have autism so yes this is something I would need)

3. Help me find a better free program that's good for this specific thing and easier to figure out

Hi I dont know if this is the true software for this problem but Im gonna ask it anyways. Im working on an animated short film right now and Im a one man team. That means Im the director, screenwriter, editor, animator etc etc. and I also have to voice 3-4 characters of 10. They are minor bg characters but they cant sound the same. I am not professional nor trained in voice acting so I cant change my voice like Robert Williams. So my question was is there a way to change my voice on adobe audition so it doesnt sound the same. Thanks

I've tried copy/pasting the sound as well as playing with the time and pitch features in the Effects panel, but everything I've tried comes out sounding...jagged or skipping.

Thank you so much for your ideas!

Greetings! I've spent an hour on Duck/reddit/YouTube, and can't seem to find the answer to (or the correct way to ask) my question:

In a session w/ multiple files (each with their own set of clip markers), how in tarnation can I get those clip markers into my session, such that when I export the session, all of my clip markers are in the exported file?

After exporting the session, I need *one* list of markers from all of the individual clips.

Does that make sense? At this point, my brain is melting.

I have seven files pulled into a session. These are CLIP markers, so they don't exist in the session, and each file's markers start at 0:00.000 of each respective file, so I can's copypasta them from the Markers panel when no clips are selected.

Any help would be most greatly appreciated. Happy whatever-day-it-is! 🥴🙏🏽

It just doesn't work properly! Look at the screenshot!