I have found that various analog modeling tools ( if you dont have analog gear)
can do a better job than chopping off peaks .

this method has to do with adding harmonics to a sound to make it more rich, rounding softly the peaks and fattening.

when mastering ,you want the transients to end up the same width and thickness throughout the frequency response.
(the sound starts off with transients of different thickness and width thorughout the frequency response)

take for instance a snare hit, the physical nature of this sound is a tiny thin pin prick transient above 8khz
a slightly wider and looser transient at 3kh
a "pop" or "crack" looser transient at 1khz to 2khz
a very loose thick transient at 300hz to 500hz
and a extreme flabby loose very wide transient at 180hz to 300hz

just a single whack of a snare drum gives us that result in the different frequencies.

by adding harmonics to the signal through analog gear or analog modelling as we thicken up the sound adding harmonics it begins to make the transients all equal and equal thickness throughout the sound wider on top where it was thin, and tighter on bottom where it was loose.
by stacking chains we can add each layer of harmonics in a way that is controlled and slightly rounds off the transient each time .
at the end of this process, there is no need for a limiter because you have an even transient throughout the frequency response, putting tape machine at end ( soft nonlinear clipping) will give you the effect of a limiter, without being destructive to the sound in any way.

all this requires that at each stage and step you actually know what you are doing and can hear what you are doing ( tiny bit at time)
in order to work without chopping peaks off in any way ( destructive toward audio)


this process is like how a vintage guitar amp works on the electric guitar transients.
multiple stages of adding slight harmonics builds of the sound and evens the transients which start off different thickness at different frequency.
each subsequent layer of harmonics added carefully, evens out the sound .

Here I give a tutorial on this process for free if you want to try it.

https://audiosex.pro/threads/the-secrets-of-music-hard-to-find-info-techniques.30049/

 

Mastering your own music:

Mastering Your Own Tracks (PDF)

 

https://www.tb-software.com/TBProAudio/ab_lmjsfx.html

AB Level Matching JSFX for Reaper




Product description
AB Level Matching JSFX matches the level of audio stream (RMS) at the beginning of your plugin chain with the level at the end. Therefore two scripts are provided: a source and a controller script.

The scripts are provided free of charge as they are, so no guarantee for correctness and error free functionality.
How to use
Just watch this video from LearnDigitalAudioTV:

Download/Installation
Please download scripts and manual here. Unpack the scripts to your Reaper JSFX folder.

System Requirements

Cockos Reaper 4.6+
or
Cockos ReaPlugs VST FX Suite 2.2
(to be used in Win DAWs with VST2 support)

References/Useful links
Reaper Forum

WaveLab Smart Bypass - like REAPER A/B equal loudness matching JSFX

 

Last edited: May 29, 2018

https://www.saintpid.se/en/isp-true-peak-limiters-test/

ISP/True Peak limiter test

ISP, Inter Sample Peak or TP, True Peak, refers to peaks in the analog domain. That is, how your peaks will look after your waveform has been converted from a digital stream within your computer, phone, iPod or CD to electrical impulses that can be amplified and played back through your speakers.



Aren’t the waveforms the same as they appear on my screen after the D/A conversion?
No, Your D/A will create a waveform that continuously moves from one sample point to the next. In that process the newly created waveform may peak above the 0dBFS ceiling which individual samples adhere to, thereby causing distortion. Just how bad this distortion depends on the converter used. In some extreme cases, the actual peak can be as much as 3-4dB above the sample peaks detected in the digital domain.

Many of the top mastering business don’t care about ISP so why should I?
No, you don’t have to treat or be aware of ISP but:

1. Digital limiters without oversampling (or other ways of detect ISP) will react to the digital waveform, not the true waveform, and thus won’t react as a limiter should react. This does not mean it will sound bad in any way, just that it doesn’t react as intended.
2. -Tip: If you want to work with limiters that lack oversampling; up-sample your 44.1 or 48KHz projects to a higher sample rate and by that have the limiter react more accurately.
3. We’ll most probably have a loudness standard for streaming media within a couple of years and that standard will be ISP aware and set at -1dBTP. This means that if your audio peak at +2dBTP it will be turned down by at least -3dB before reaching the end user. All that extra loudness you gained by letting random equipment create a positive peak (that might distort) will be lost.
4. Encoding to lossy formats; ISP will make it harder for encoders to do a good job. If you don’t have the tools or knowledge to check how your audio will perform post-encoding I would recommend to stay away from positive True Peaks.

Since it’s more or less guess-work, or at least really hard to create a real-time limiter that handles ISP perfectly, we thought we’d put as many limiters as possible to the test. So a handful of mastering colleagues and I (Ian Stewart, Sigurdór Guðmundsson, and Johan Eckerblad) went to work.

The Test

We used a mastered track that we boosted by 7.5dB (the peaks were at -1dBTP/dBFS in the mastered file so max Gain Reduction would be 6.5dB), we set the limiter ceiling to -1.0dBFS (or -1dBTP if available in the plugin), rendered it as 44.1KHz, 32bfp wav and measured the True Peaks of the resulting waveforms. Limiters with a result as close to -1.0 dBTP as possible have handled ISPs the best, whereas values over -0.8 dBTP (shown with increasing values in white, through yellow, to red when they exceed 0 dBTP) mean that the limiter has failed to handle ISPs to within the margin of error for measurement we observed during testing (more on that in a bit).

Here’s the track that we used:

• “Yes” or “No” in the TRUE PEAK column indicates whether the developer mentions or claims that the plugin handles True Peak or ISP, either in the manual or marketing. A “No” does not necessarily mean that it doesn’t handle or try to handle ISP, just that it’s not explicitly stated by the developer.
• The results published here are measurements taken using Izotope RX. Additionally, we did measure with Sequoia, Wavelab, and Nugen, and even though we observed slightly different results, they were negligible. The biggest differences were in iZotope’s products. Since they probably use the same detection algorithms in all their software, this was expected. The other software measured within a 0.1 to 0.2 dB difference, which still puts iZotope amongst the best performers.
• Where other settings such as Attack and Release were present we used either the default or adjusted it to a value as close to standard as possible
• This test did not take into account sonic qualities at all. That will have to wait for our planned podcast or another post.
• It’s not recommended to use any of the limiters with a positive score in 44.1KHz projects. They should handle ISP better as your sample rate goes up.
• If your favorite Mastering Limiter ain’t here, send us a link so that we can try it out. However, we’re not in the business of buying every limiter out there, so any you’d like us to try must have fully working demo-versions.

https://docs.google.com/spreadsheet...13siVO2VWSHf9abJKyPL_LSNE/preview/sheet?gid=0

 

Last edited: Jul 3, 2018

https://www.saintpid.se/en/checklist-before-mastering/

Checklist before you send a track for mastering
By following this guideline you give us the best conditions to deliver a great sounding master:

• Remove all Limiters, compressors, EQ or other processes from your master bus. If you think a master bus process is necessary, send two versions and explain why you think the process is necessary, name the files accordingly (for example cool_things_-_clean_mix.wav and cool_things_-_bus_compressor.wav)
• Render your material with the same sample-rate used while recording.
• Deliver 24bit files or higher, preferable is a 32bit floating point file if your DAW supports it.
• Mix your track with lots of headroom, leave at least 3dB but we won’t complain if we get 6-8dB of unused bits. If your mix already pushing 0dB, do not just lower your master volume. Instead, decrease the gain of each track to create headroom.
• Render your track so that you have about a second or two of silence both before the music start and after the last note has faded out, that way you’ll be sure nothing gets lost.
• Give us one or more reference tracks. A Spotify or other streaming link will do but where it’s possible we prefer downloadable files.
• What format do you need? Check with your distributor/label what they prefer, format, sample frequency, and bit depth.
• Do you have all info that’s needed for a final master? For example ISRC and EAN/UPC for CD, track order, all the correct names of artist and tracks?