fradoca
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I've discovered this new free vst plugin(both mac and pc). I use it for audio mastering with wavelab and for music reproduction with jriver and windows fidelizer with my audeze lcd-x and audio gd master 9.
Use it with native high-res material or upsampled red-book stuff and you'll hear the benefits both sonically and musically.Well worth a try!
Here's the link :
https://vladgsound.wordpress.com/2014/12/21/tdr-ultrasonic-filter-alpha-version/
here are some comments :
"I’ve tested using this on several 96k mixes which include somewhat long fx chains, inserting instances between every effect. I’ve been setting them to be more subtle/higher cutoff on the master buss, moderate on busses, and deeper on individual instruments. The results have been noticeable indeed! I am discovering extra clarity in the upper mids and less overall muddiness. It may seem counter-intuitive to increase the clarity of the upper part of the spectrum by removing part of the upper spectrum, but it absolutely works. I am also noticing, which i did not expect, that the stereo field is being affected as well… on one mix which was a bit narrow it seems to add width, and on another mix which was a bit wide it seemed to solidify the center. Further testing on my part is needed, but following this process seems to subtly “automatically” balance the stereo width in a pleasing fashion."
"I experimented with the Ultrasonic filter on a just-for-fun master at 96kHz. I had about 10 plugins in a chain (all producing harmonic distortion of various degrees, mainly Acustica Audio Aqua and Nebula instances) and smashed into Voxengo Elephant ridiculously hot (limiting up to 8dB at best). I inserted Ultrasonic filter first in the chain and then after (and thus before) each and every plugin, including after elephant (so actually after the limiting.. just before final dither). The difference is ASTONISHING!! It is so ridiculously obvious and much better with the filters than without. A truly mind-blowing “What???!!” experience for me. It basically removed a lot of problems I’ve identified in mixes and previous masters that had heavy processing with a lot of harmonics.. mainly that nasty build up of “ringing” around 2.3 to 4kHz area. It also cleared out the congested mids and lower mids.This is a game changer for me. I will not do any more high resolution mastering without this filtering. Period. Heck, I will never mix at high resolution again without this plugin inserted all over the place"
if you have a player that supports vst plugins use it after the resampler.
Here are my settings :
freq: 24 khz
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slope : 25%
amount : 100%
i can tell you that since i've discovered it i use it all over the place.

dmbr
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Hm JRiver's resampler is built in...how might one implement this VST with it?
I'm also a little confused as to what it does still
I'm also a little confused as to what it does still

dmbr
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Hm, doesn't seem to work with JRiver--displays the message "System bandwidth close to audible bandwidth. Filter bypassed" 
dmbr
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Anyone know what might be wrong?
dmbr
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I figured it out.
This VST is exclusively meant for 96khz+ files, and is bypassed if fed anything less.
Sadly, JRiver's resampler can't be moved from the front of the chain, which means there's no using this VST in JRiver if you want to output below 96khz
I use Out of Your Head and it requires a sampling rate of 48khz, so no dice.
I did, however, get it to work with Foobar (had to put it at the very start of my long DSP chain). And wow...it really does improve SQ noticeably! Greater clarity and separation, and the mids "bloom" a bit. Love it.
Problem is, I prefer JRiver
its channel up sampling works better with OOYH than anything available to Foobar.
So...does anyone know of a way to resample to 48khz *after* applying a VST in JRiver?
This VST is exclusively meant for 96khz+ files, and is bypassed if fed anything less.
Sadly, JRiver's resampler can't be moved from the front of the chain, which means there's no using this VST in JRiver if you want to output below 96khz
I use Out of Your Head and it requires a sampling rate of 48khz, so no dice.
I did, however, get it to work with Foobar (had to put it at the very start of my long DSP chain). And wow...it really does improve SQ noticeably! Greater clarity and separation, and the mids "bloom" a bit. Love it.
Problem is, I prefer JRiver
So...does anyone know of a way to resample to 48khz *after* applying a VST in JRiver?
iano
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I am no expert on JRiver, but I am wondering if all else fails whether you can do a two/three step process...i.e. do the upsampling, then apply the filter then downsample - looks like only one change in output format can be done in one go in JRiver as far as I can see...anyway, that's what I would be tempted to do...its a pain to go through separate steps I realize, and maybe someone else knows of a better way - but if you wish to use JRiver for upsampling, and OOYH for the ultimate product, and use the filter, then at least this more laborious series of steps may get you there. And you can apply this to many tracks at the same time and give it some time to run.
OOI I use sox...but I did recheck on http://src.infinitewave.ca/ looking at a comparison with sox (very high quality with aliasing enabled) and JRiver (v18) resampler, and indeed there seems to be some a few improved measurements with JRiver (I do not know if they would be audible)...so I may even try the JRiver 3 step process myself
I would like to thank the OP for starting this thread...just started using the filter on my upsampled tracks and I think I notice a positive difference - I read through the various gearslutz forum discussions to get a background, downloaded the files mentioned there and on the plugin's site.
I know I should probably go through the ABX process to 'prove' its not just the placebo effect...but to be honest I cannot really bother - the up-sampling and applying the filter is really simple for me to apply to many tracks at a time...I would rather listen to the music
OOI I use sox...but I did recheck on http://src.infinitewave.ca/ looking at a comparison with sox (very high quality with aliasing enabled) and JRiver (v18) resampler, and indeed there seems to be some a few improved measurements with JRiver (I do not know if they would be audible)...so I may even try the JRiver 3 step process myself

I would like to thank the OP for starting this thread...just started using the filter on my upsampled tracks and I think I notice a positive difference - I read through the various gearslutz forum discussions to get a background, downloaded the files mentioned there and on the plugin's site.
I know I should probably go through the ABX process to 'prove' its not just the placebo effect...but to be honest I cannot really bother - the up-sampling and applying the filter is really simple for me to apply to many tracks at a time...I would rather listen to the music

dmbr
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For now I'm settling for using DirectSound instead of WASAPI output to OOYH in order to get the VST working.
I set JRiver to resample 44.1 content to 48, and for 96khz+ content to be left alone by JRiver and resampled to 48 for OOYH by Windows.
I need to do a little more listening to see if losing bit perfect output is worth the subtle improvements this filter can offer (I also need to play with the settings more). It's a painful decision :/
I wonder why JRiver doesn't allow you to move the resampler up/down the DSP chain?
I set JRiver to resample 44.1 content to 48, and for 96khz+ content to be left alone by JRiver and resampled to 48 for OOYH by Windows.
I need to do a little more listening to see if losing bit perfect output is worth the subtle improvements this filter can offer (I also need to play with the settings more). It's a painful decision :/
I wonder why JRiver doesn't allow you to move the resampler up/down the DSP chain?
fradoca
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thanks for all your nice comments.It seems that my sonic impressions about the filter are the same as yours.
I use it with jriver and i upsample everything to 192khz with the internal jriver resampler then i apply the filter during the playback.
Also remember to check the first box in the dsp section of jriver : output format.
that one enables the internal 64 bit processing and that makes really a difference in sound when compared to foobar.
Also disable any internal volume control and set flat line overflows instead of clip protection.That will even enhance the performance.
I use it with jriver and i upsample everything to 192khz with the internal jriver resampler then i apply the filter during the playback.
Also remember to check the first box in the dsp section of jriver : output format.
that one enables the internal 64 bit processing and that makes really a difference in sound when compared to foobar.
Also disable any internal volume control and set flat line overflows instead of clip protection.That will even enhance the performance.
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