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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: jes1982 on March 10, 2024, 12:27:21 AM
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Hi-
As the Subject says, I'm having an issue with the sample type conversion and am seeking guidance. Hopefully someone can educate me about this problem.
I'm working with 24/48 WAV files in the latest version of Audition. I used the Hard Limiter (screenshot attached of settings) to boost the gain with the maximum amplitude set at -0.1db. No problems. I then used Audition's Sample Type converter to save a 16/44 file set. Again, screenshot attached of the settings I used. Unfortunately, the conversions introduced clipping and I'm not sure why as the there is no clipping or distortion in the 24/48 files.
Some troubleshooting I attempted: (1) used a different setting for the sample type conversion (i.e., no dithering) and that didn't change the result; (2) forwarded the 24/48 file set to a friend who also has Audition (a different version, I think) and he didn't fare better; and (3) I converted the files with the latest free version of r8brain and that also introduced clipping.
Any ideas on what's going on here? I will add that I pushed the gain levels more than usual, but, again, there's no clipping or distortion in the 24 bit file set. Naturally, I do all the editing with 24 bit files before converting to 16/44.
Thanks for any counsel you can provide!
P.S. The workaround is I can run the Hard Limiter a second time with the 16 bit files to eliminate the clipping, but that's more work.
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The first thing I would try is converting the file before doing the normalization to see what happens, if you still have a version of the file before that was done.
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My solution - don't convert to 16/44. It has been 7 or 8 years since I made 16/44 files. I simply circulate my 24/48 files.
Also, I never had clipping introduced when I resampled/dithered a file in Wavelab 6, so not sure. I also generally only increased the amplitude to -0.2 dB as some playback introduced artifact as you got closer to 0.0 even if it wasn't over.
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The recommendation for files that will become mp3’s is to normalize or master to -1dBFS. No higher. I’ve noted in mastering things for release that it’s common to see peak increases of +0.2 from sample rate conversion, occasionally as much as +0.8.
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The recommendation for files that will become mp3’s is to normalize or master to -1dBFS. No higher. I’ve noted in mastering things for release that it’s common to see peak increases of +0.2 from sample rate conversion, occasionally as much as +0.8.
I've never had an issue at -0.2 dB
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My solution - don't convert to 16/44. It has been 7 or 8 years since I made 16/44 files. I simply circulate my 24/48 files.
Thanks for the input. Just to clarify: I also archive 24/48 FLACs. My standard practice now is to save (1) my 24/48 "raw" file; (2) edited 24/48 FLACs (I won't say "mastered" because I don't know what I'm doing); and (3) 16/44.1 FLACs. Overkill, I guess. I'm in the process of trying to clean out my parents' house and cursing the things that weren't thrown away. When my son has to deal with my hard drives, if he looks at what's there, he may wonder why his father had so many damn copies of the same show.
But, yes, the audience for the music I tape prefers the redbook version and I'm going to go along with that for now. I suspect it's because the files take up less space, not a desire to burn to CD-R.
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The recommendation for files that will become mp3’s is to normalize or master to -1dBFS. No higher. I’ve noted in mastering things for release that it’s common to see peak increases of +0.2 from sample rate conversion, occasionally as much as +0.8.
I've never had an issue at -0.2 dB
Thanks for the comments. This is interesting and gave me an idea. For one of the files I was having trouble with where I had initially set the maximum amplitude at -0.1db, I increased the hard limiter by 0.1db increments. I didn't get a clean conversion until the maximum amplitude was set at -0.5db. This means the sample type conversion boosted the the left channel by +0.41db and the right channel by +0.29db.
Now I wonder: if I can find or create a "hotter" file for testing purposes, will the sample type conversion boost the gain even more than +0.41db?
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Now I wonder: if I can find or create a "hotter" file for testing purposes, will the sample type conversion boost the gain even more than +0.41db?
Huh, interesting. I've long been normalizing to -0.5dB without issue, but I'm now wondering if it wouldn't hurt to go a bit lower.
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I use Sound Forge and normalize to 98.14% (not sure what the dB value of that is), and then dither with iZotope's mbit process. I never noticed any clipping after dithering.
Like Rory, I've now stopped offering 16 bit files (unless the source was 16 bit). I haven't had any complaints so far, or requests for a 16 bit version. So either most people are accepting 24 bit, are dithering the files themselves, or just don't bother downloading.
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I use Sound Forge and normalize to 98.14% (not sure what the dB value of that is), and then dither with iZotope's mbit process. I never noticed any clipping after dithering.
Looks like 98.14% is about -0.16dB.
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I use Sound Forge and normalize to 98.14% (not sure what the dB value of that is), and then dither with iZotope's mbit process. I never noticed any clipping after dithering.
Looks like 98.14% is about -0.16dB.
I was just coming to post that! :D
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I don't know if this helps or not, but I use Audition as well (the latest version for Mac: Audition 2024 build 24.2.0.83). I always normalize to -1 dB and then use the "File > Export" dialog to create 16/44.1 files. Never any clipping.
Actually, I kill several birds with one stone by normalizing, then listening back while reducing noise and placing markers, then exporting the marker ranges (individual tracks) at 16/44.1. I started doing this when I could no longer use CDWave and it works well. I previously used three pieces of software (Audition, r8brain, CDWave) to do the same thing I now do with only one.
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The recommendation for files that will become mp3’s is to normalize or master to -1dBFS. No higher. I’ve noted in mastering things for release that it’s common to see peak increases of +0.2 from sample rate conversion, occasionally as much as +0.8.
I'm in the -1dbFS camp with Aaron and EmRR. Sufficiently normalized, never a problem. Safe and easy.
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Sample Rate Conversion (SRC) can introduce peak levels beyond 0 dbfs due to interpolation. So the order of operations is the key.
Gordon
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Yes, the bitdepth reduction last.
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Sample Rate Conversion (SRC) can introduce peak levels beyond 0 dbfs due to interpolation. So the order of operations is the key.
Gordon
there was a huge debate about this
“Intersample overs” back in the day when benchmark introduced the DAC 2. While I generally find the technical discussions of John Siau to be impenetrable, the “community” as it were just poo-poo’d the idea as if it wasn’t a thing, or at least not as significant as benchmark made it out to be
https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings
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I use Sound Forge and normalize to 98.14% (not sure what the dB value of that is), and then dither with iZotope's mbit process. I never noticed any clipping after dithering.
Looks like 98.14% is about -0.16dB.
I was just coming to post that! :D
I'm curious as to the math behind y'all coming up with that number. Care to enlighten me?
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I'm curious as to the math behind y'all coming up with that number. Care to enlighten me?
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FWIW, the sharpest mastering engineer that I know says -3db, and why not -6db for 24 bit?
Let those peaks breathe!
Pop music be damned
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Goldilocks middle way. Not too much, not too little. No need to overcompensate when normalizing by trying to get it as close to 0dBFS as possible. Just need enough extra space so the complete waveform doesn't get chopped off at the top.
If peaks are way down then normalize to a peak level closer to 0dBFS. If unsure exactly how high that peak actually goes because it might happen to crest in between two sample points (an intersample over) choose a slightly lower target peak value. That's it. One or two dB down from 0 is plenty and equates to a more or less insignificant difference in the position of the listeners playback volume knob.
-3dBFS is not unreasonable. ..and that or even -6dB is likely what a mastering engineer would prefer to receive from a client. But a target of -6dBFS as a target for final release gains nothing and will require additional playback gain to compensate, as it will most likely sound significantly quieter than whatever is played before or after. The additional 3dB of empty space gains nothing. -6dBFS is totally fine as a "released as it was recorded peak level" that isn't so low it really needs normalization to be listenable, yet is an overly low target for final normalization of a finished recording where levels are being actively adjusted in post production.
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I do not normalize, I add gain up to -0.2dB to my final wav 24/48 files.
I *may* convert down to 16/44.1 for some odd reason, but gave that up as a workflow habit many years ago.
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I do not normalize, I add gain up to -0.2dB to my final wav 24/48 files.
Isn't "adding gain up to X dB" exactly what normalizing is?
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it is. Was just typing the reply below..
I do not normalize, I add gain up to -0.2dB to my final wav 24/48 files.
[..snip]
That is normalizing. What you describe is the process you use to achieve the desired normalization of level.
There are various ways to do it, which including tools that automatically calculate how much gain to add to reach the desired target level. Complicating things is the often unstated "level of what?". We're talking about peak levels not being overly low nor exceeding 0db Full Scale. But level normalization could also be referring to loudness normalization, which is a perceptual measure that relates more to RMS value normalization or some measure other than "peak".
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I do not normalize, I add gain up to -0.2dB to my final wav 24/48 files.
Isn't "adding gain up to X dB" exactly what normalizing is?
YES and NO. >:D Not "exactly".
Depends on which type of normalization.
https://www.izotope.com/en/learn/audio-normalization.html
Peak vs. Loudness Normalization:
Peak Normalization: Focuses on adjusting the gain to bring the loudest peak of the audio to a target level.
Loudness Normalization: Adjusts the gain to bring the average loudness of the audio to a target level, often measured in LUFS (Loudness Units relative to Full Scale).
So given all that, I prefer to do it "manually" so to speak.