Old 03-25-2020, 05:09 PM   #1
phoo
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Default Inharmonic or Intermodulation Distortion Issues

I have a very complex mix in the works. It has a lot of high frequencies that are much like sibilances, but not all are from voices. Some are "airiness". Yes, these are up there, and yes they have been EQed as much as would normally be done. The problem is not tone, nor them occupying the same space. This has turned onto a technical issue. Compromise is expected, but I'm trying to understand why this is happening.

The problem is when some tracks are mixed there is what sounds like intermodulation distortion being produced to the point it sounds like inharmonic distortion. Two clean smooth tracks mix and sound like they are being run through a clipper. The sound is grainy at best, and has crackles at worst.

Tracks are 24 bit float, playing as 32 bits - 44.1K sample rate. Mixing to the same.

This is not a level issue as all tracks are well below any kind of clipping. The mixed tracks are around -12db. For that matter, since mixing is done in 64 bits I don't see how it would be possible to clip during the mix.

I'm guessing intermodulation, which will create frequencies not present in any of the source. This could and would cause what I'm hearing. Lower "new" frequencies would sound grainy. Higher "new" frequencies could and would cause terrible distortion is those frequencies are above the nyquist frequency. I can justify (via my ignorance) that this is what's happening.

It's also similar to dropping to 8 bits, or lower.

Solutions? Well, yes, EQ out the HF of the sources until it goes away. That's impractical, as I'm having to EQ out everything above 5k or so for it to be completely avoided. That's OK for some tracks, but completely ruins the sound of others, but to cut off the cymbals at 5K, or vocal sibilance? Not really.

Solutions that seem to have an effect, after EQ had failed is quite interesting. These have worked with a good bit of success. Each causes a good number of questions to arise. These don't work at all on some tracks and with more than two tracks isn't much more complicated when any pair of tracks is OK.

Solution:

1) Flip the phase of one track.

2) Shift the timing of a track a few msecs, or samples.

3) Use the RePitch plug-in to change the pitch just a little, or just the formant.

Think of this problem as being "how to mix multiple tracks of white noise" in such a way as to not create the artifacts that intermodulation can cause.

In the real world this is happening when the snare hit, there is a cymbal crash, or there sibilances, all while a smooth airy mellotron is sustaining chords.

What is actually causing this, and are there better solutions to avoid it?

I know this is a lot to read, but I hope I explained the problem well enough for discussion.

THANKS.

(please forgive any typos)
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Old 03-25-2020, 05:25 PM   #2
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In one of these smooth mixes, what happens when you go down the mixer and mute tracks, both one at a time and cumulatively? And in various strategic combinations?
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Old 03-25-2020, 06:05 PM   #3
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Quote:
Originally Posted by phoo View Post
I'm guessing intermodulation, which will create frequencies not present in any of the source.
That's true, but inter-modulation distortion requires a non-linearity to be applied to the sum of the signals.
Do you have a non-linear plugin on a bus or the master? If you don't it can't be inter-modulation distortion.
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Old 03-25-2020, 06:15 PM   #4
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-What application you're using to play those files "as 32-bit float" (VLC or other media player?)

-Did you record the files as 24-bit float in Reaper (is that even a possible driver setting in Reaper?),

-What audio hardware you have,

-What your driver settings are (in Reaper especially).

THANKS! Good points, all of them.

I misspoke about the format being float. It is 44.1khz, 24 Bit PCM. The default rendering format of Reaper, and the recording format.

I find the combinations of tracks by muting tracks one at a time until the problem goes away, then it's bring back tracks one at a time until it shows back up. It's not always the obvious combinations.

I have no idea is the plug-ins are non-linear. All are Reaper installed by default. ReaEQ mostly, to roll off the lows and highs to limit bandwidth. That said, the issue it there with no plug-ins in the path at all. Since this should rule out intermodulation, what's the remaining possibilities?

All individual tracks were recorded in Reaper. Audio interface is a ZOOM UAC-8 (USB3) via a main recording desktop, Windows 10 (sigh), and onboard audio in an HP Envy Laptop. The artifacts happen on both machines. Record on the desktop, then the majority of post recording work is done on the laptop. Everything is then passed back to the desktop for final "real" mixing through good speakers and final rendering. (I haven't noticed any differences between rendering on either machine if that matters - all that should stay in Reaper software)

Playback of the rendered files is in Reaper, VLC, Adobe Audition, and ultimately in Sony CD Architect, and in the car via blutooth directly from the Reaper on the laptop. (love mixing in the car - grin number one)

Track mixing in Reaper is 64 bit float.

The artifacts are audible during realtime playback in Reaper on both machines. What's being heard is what's rendered. This definitely makes it much easier to isolate. Because of this it seems like it would be irrelevant what the rendered format is, unless Reaper changes it's realtime playback amd mixing based on the rendering setting.

It also seems to not be computer or hardware specific since it happens on both the laptop (with really crappy audio) and the desktop (much better and drastically different hardware). Also, the fact that it's audible in every place I've tried to play these on may also help rule out it being something app specific. It's happing in Reaper, and it's not in any of the individual tracks.

Yes, I do understand how downsampling for playback could be relevant. Some apps, such as CD Architect has a dithering plug-in enabled by default, specifically for burning to 16 bit CD. That does make playback sound better, but it can't remove pops and clicks in the source (which is ultimately how this ends up sounding in the song unfortunately). At least there's no samplerate conversions going on.

These artifacts are subtle. My wife can't hear them at all. Unfortunately, I've "heard them" and now they are all I can hear. (grin two)

(I'll edit this response with the driver settings shortly. Not where the desktop is at the moment.)

Thanks for the responses!
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Old 03-25-2020, 08:56 PM   #5
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Since it seems to affect the highs specifically, there is a fair chance you are running into aliasing in plugins. One solution is to change the project sample rate to 48k or even 88.2k.

Aside from this, intermod distortion can be caused by any nonlinear processing that is fed more than 1 frequency at a time. compressors, limiters, saturators, clippers, even clipped samples.

Start by putting a spectrum analyzer on the master track. Solo each track, one at a time, and look for audio frequencies that aren't a multiple of the fundamental note. Those are the culprits. If these only show up when 2 or more tracks are soloed, it is caused by intermod distortion.
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Old 03-26-2020, 05:07 PM   #6
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Quote:
Originally Posted by Philbo King View Post
Since it seems to affect the highs specifically, there is a fair chance you are running into aliasing in plugins. One solution is to change the project sample rate to 48k or even 88.2k.

Aside from this, intermod distortion can be caused by any nonlinear processing that is fed more than 1 frequency at a time. compressors, limiters, saturators, clippers, even clipped samples.

Start by putting a spectrum analyzer on the master track. Solo each track, one at a time, and look for audio frequencies that aren't a multiple of the fundamental note. Those are the culprits. If these only show up when 2 or more tracks are soloed, it is caused by intermod distortion.
Thanks!!! I believe you may have hit on the most likely culprit with the aliasing. The tracks in question all have a good amount of high harmonics, as mentioned earlier. It's the combinations of those in very specific places.

I've done some experimenting with raising the samplerates of the projects and done some rendering, but haven't done enough to see if the problem is affected for the better. One step at a time - changing the project first, and making sure there are no obvious differences because of that that. So far none.

I opened the files rendered at higher samplerate in Audition and looked at the spectrum in there. I can see hints of energy above 22K in them. That to me does seem to point to there potentially being some energy above the nyquist frequency. Audition won't display anything above that for any given samplerate (grin - well duh).

Thanks for your thoughts. I'll post here if I find something definitive. I suppose others have run into this, so I this helps them. That said, my guess is most folks won't be mixing projects with this much HF. Airy mellotrons are special.
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Old 03-30-2020, 07:05 PM   #7
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Big BINGO on the aliasing. After converting the project in question to 88.2khz and rendering out, and opening it in Audition, the HF showed up in all its glory.

The screenshot of Adobe Audio in spectrum view shows a lot of energy above the nyquist frequency of 44.1khz. There also appears to be some (maybe) dithering up around 35khz, even though dithering in the rendered mix was not enabled. This is straight out of Reaper.



Converting the project after the fact didn't get rid of all the artifacts, unfortunately. Some audible aliasing artifacts are in a few the original wav files, as those are 44.1k. Can't fix these with filtering or resampling, but I can go back and rerender a few as 88.2khz. Most of these are from a couple of mellotron and synth VSTs. I still have the originals in Reaper, having rendered them to wav for the main project.

While experimenting with this I discovered that the main desktop recording computer can't play the project at 88.2hz because it hits the CPU way too hard. The desktop does support it, as I have recorded a few tracks at that sample rate this week. There is just too much going on, resampling of the underlying wav files probably. That said, the laptop is happy playing at 88.2khz. The onboard audio, as crappy as it is, supports that playback sample rate and the CPU is indeed more powerful than the desktop and has more RAM. Might be time to start thinking about a replacement for the desktop. The motherboard is from 2004. It's still doing great at 44.1k, and even higher sample rates if there aren't many tracks. Because of all that, I have to render on the laptop, but have to mix at 44.1khz on the desktop for the good speakers. It's a compromise I must learn to live with.

But, now I know what the problem is and can easily avoid it in future projects.

Thanks everyone for your help, thoughts, questions, and answers.
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