This comes up constantly, and the answers online are usually either a marketing page or a forum argument from 2014.
This covers standalone audio tools 2.0 Workflow the way it actually works in a session: the mechanism, the numbers that matter, and the decisions that follow from them.
Nothing below is theory for its own sake. Every number here is one you can check on a meter, and every step is one you can run on the next track you open.
Key takeaways
- Process from a lossless source, and convert to lossy once, at the very end.
- 24-bit gives 144 dB of dynamic range and is the working standard.
- Check whether the tool normalises output, because that changes your gain staging.
- Keep the unprocessed original, always.
What the tool does, precisely
Online audio tools have a reputation for being toys, and a lot of them are. The difference is whether the processing happens at full resolution on the original file, or on a downsampled preview that gets re-encoded on the way out.
Processing that matters happens at the source sample rate and bit depth, writes a lossless result, and does not silently normalise or re-encode your file on the way through.
- Work from the highest-quality source you have, not an MP3 of the thing you want to process.
- Keep the output lossless if any further processing is coming.
- Only convert to MP3 or AAC at the very end, once.
- Check the output level. Many tools normalise by default, which quietly changes your gain staging.
File formats, and when each one is correct
| Format | Type | Use it for |
|---|---|---|
| WAV | Uncompressed PCM | Mastering, delivery, archiving, anything with more processing to come |
| FLAC | Lossless compressed | Archiving and distribution where size matters and quality cannot |
| AIFF | Uncompressed PCM | The same job as WAV, historically on Apple systems |
| MP3 | Lossy | Final listening only. Never an input to processing |
| AAC | Lossy | Streaming delivery. Better than MP3 at the same bitrate |
| Bit depth | Dynamic range | Where it belongs |
|---|---|---|
| 16-bit | 96 dB | CD, and the final delivery for physical media |
| 24-bit | 144 dB | The working standard for recording, mixing and master delivery |
| 32-bit float | Effectively unlimited | Internal DAW processing and field recorders that cannot clip |
| Sample rate | Nyquist limit | Where it belongs |
|---|---|---|
| 44.1 kHz | 22.05 kHz | CD and the majority of streaming delivery |
| 48 kHz | 24 kHz | Video, broadcast and games |
| 96 kHz | 48 kHz | Hi-res delivery and heavy pitch or time processing |
Getting a clean result
- Upload lossless if you can. Every lossy generation compounds.
- Do destructive edits once, at the end, from the original.
- Keep the unprocessed file. Always.
- Listen to the output against the input, level-matched.
- Check the top end for encoder artefacts if the source was ever lossy.
Sample rate conversion quality varies enormously between tools. A poor converter introduces aliasing that no later processing removes.
How it translates on real systems
Almost nobody hears your record on the system you made it on. The point of every decision above is that it survives the journey.
| System | What it exposes | What to check |
|---|---|---|
| Phone speaker | No low end at all below roughly 500 Hz | Does the bass line still read from its harmonics? |
| Earbuds | Exaggerated width and sub, hyped top | Is anything sibilant or fatiguing? |
| Laptop | Thin midrange-only playback | Is the vocal still intelligible? |
| Car | Boomy low end and road noise | Does the low end turn to mush at 80 to 120 Hz? |
| Club system | Everything below 40 Hz, loud | Is the sub mono and controlled? |
If it works on a phone and in a car, it works almost everywhere else. Those two are the honest tests.
A ten-minute version of this
If you have one evening and not one week, this is the order that gets the most improvement for the least time.
- Two minutes: listen all the way through without touching anything, and write down the three things that bother you.
- Two minutes: fix the loudest problem on the list, with the simplest tool that will do it.
- Two minutes: check in mono and on a phone speaker. Fix anything that falls apart.
- Two minutes: level-match against a reference and note the tonal difference, not the loudness difference.
- Two minutes: make one broad corrective move based on that comparison, then stop.
Three specific problems solved beats twenty small adjustments that cancel out. The written list is what stops the session drifting.
Frequently asked questions
What is the most important thing to get right with standalone audio tools 2.0 Workflow?
Tonal balance and translation. A track that holds up on a phone, in a car and on monitors is finished. One that only works on your monitors is not.
What loudness should I target?
-14 LUFS integrated with a -1 dBTP ceiling covers almost all streaming. Go quieter for acoustic and classical material, louder only where the genre genuinely demands it.
How much headroom should the mix have?
Three to six decibels, with no limiter on the mix bus. That gives the mastering stage something to work with.
How do I know when it is finished?
When it holds up level-matched against a commercial reference, translates on several systems, keeps its low end in mono, and your changes have stopped improving it.
Hear it on your own track
Upload a mix and get a mastered version back in minutes. Free to try, no card required.
Related tools: Audio Trimmer · Sample Rate Converter · BPM Finder
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