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Topic: Room Correction Hardware/Software Issues, Options. Please Advise. (Read 2085 times) previous topic - next topic
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Room Correction Hardware/Software Issues, Options. Please Advise.

Please excuse this lengthy post but I didn't want to forget asking every question I had about this complex topic, and if I'm going about solving my problems in the best sounding way.

Media players like roon rightly have their adherents especially if it may offer seamless solutions for room correction.
https://community.roonlabs.com/t/solution-for-4-channel-motu-m4-or-focusrite-audio-interface-and-xo-in-a-f-a-s-t-configuration/139931

However, I don't like being tied to one particular player-be it Foobar, JRiver, VLC or my old version of Samplitude DAW.

Presently, my bigger problem is that my custom designed main speakers are still away from being built, I haven't time yet to shop for and settle on a DAC (s) and have no hands on experience using Camilla, DIRAC Live, REW or any other room correction software.

And based on the following questions I won't yet know which if any RC software can best help me correct any glaring room mode problems, beyond what I will first need learn to correct acoustically (as all experts advise).

Then how to compensate for however much gain loss the software may cause, where the DACs' output voltages might be inadequate to cleanly drive my main speakers power amp and/or the plate amps of my four Rythmik F12 subs.

Additionally, while these MCH DACs likely sound amazing at least for the price https://www.oktoresearch.com/dac8pro.htm https://motu.com/products/avb/8a https://beta.prismsound.com/products/titan/ , how good will their tonality and other subjective sound quality metrics (e.g. soundstage size, imaging) be for use in the entire system? That is, should I instead use one of these MCH DACs for my subs only and use a perhaps better sounding stereo DAC for the main speakers? https://www.kitsunehifi.com/product/holo-audio-may-dac/

I'm thinking that this may be advantageous because my main speakers and subs are passively crossed and because DACs chips (AKM) used in some stereo DACs may be easier to configure and also use with output stages to help reduce intersample overs distortion, which unfortunately is often a direct consequence of the "loudness wars" between competing record producers and/or artists. https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings?_pos=1&_sid=0eeb1f150&_ss=r
 
Much discussion about that ongoing tragedy here https://gearspace.com/board/mastering-forum/1401406-intersample-clipping-audible.html

At the same time at least some of those RC software and/or DAC choices may also depend on the following

First, my passively crossed mains only play down to 70Hz, though I suspect that any of those three MCH DACs-perhaps especially the Okto (a good choice even though unlike the other two it has no ADCs??) will reproduce that range from ~ 65Hz on down quite faithfully. Yes?

Second, would there be audible or disastrous timing issues if two USB ports from my pc (music source) each feed the USB input on the stereo DAC and MCH DAC? If yes, can it be solved by simply connecting one of the PC's USB ports to a USB hub to thereby feeding both DACs from the same USB port? If neither of those fixes work then what might?

Third, would there be any benefit to using the "better" sounding stereo DAC for the main speakers and the MCH DAC for the subs but to only have the subs' DAC in the "convolving" filter loop, since it's the subs which mostly interact with the room's mode (e.g. standing waves). Yes?

Fourth, as most MCH DACs are made for the pro audio community they include an ADC for each channel, where consumer DACs like the Okto don't have any built in ADCs. Obviously, this would require that a separate ADC would needed be used between the PC's USB input and the test signal recording microphone (and if need be a mic preamp) for the live recording of each test tone playback by a speaker and sub. So what model ADC and mic to use? 

Fifth, which if any RC software flavor might likely impose the least amount of gain loss to avoid needing line stages between the DACs' outputs and the mains power amps or subs' plate amps?

Ultimately, what worries me is that using two different DACs will make part of what's heard sound colored, but I can't know for sure. And then how problematic are the other questions. Please advise. 


Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #1
For audio convolution I use only ffmpeg. Supports both float and double sample formats. And input to any output channel mixing, and many other features.

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #2
Not your main item, but this is solvable:
I'm thinking that this may be advantageous because my main speakers and subs are passively crossed and because DACs chips (AKM) used in some stereo DACs may be easier to configure and also use with output stages to help reduce intersample overs distortion, which unfortunately is often a direct consequence of the "loudness wars" between competing record producers and/or artists. https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings?_pos=1&_sid=0eeb1f150&_ss=r
 
Much discussion about that ongoing tragedy here https://gearspace.com/board/mastering-forum/1401406-intersample-clipping-audible.html
Intersample overshoots "are only a problem near full volume". If your playback application (digitally) attenuates, you have more headroom.
Solution: Play from files, with ReplayGain. Scan and tag your files, and use a ReplayGain-aware player. More securely, use a ReplayGain scanner with upsampling, which measures (to the capability of the algorithm) the true intersample peak and sets the peak tag accordingly.
foobar2000 can do that, and also limit output volume according to peak to keep you safe even when you run the player's volume slider to full.

It has been discussed in-depth at this forum too, most recently at https://hydrogenaud.io/index.php/topic,124977.0.html . In most cases, the RG gaini figure will attenuate enough. But if you set playback to be limited according to the peak tag - after scanning as above - then the problem is solved.

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #3
USB has no role in syncing two separate ADC and DAC, you at least need to chain them using SPDIF/AES3/ADAT for synchronization. Also, only ADAT is capable of multiple (eight) LPCM channels, SPDIF and AES3 only support stereo, and ADAT only supports up to 48kHz when using all 8 channels. Syncing can be achieved by setting one of the device as master and another one as slave, or via SRC if supported by the device's hardware. However such methods are rather cumbersome and more prone to drop out, so just get an all-in-one interface (ADC + DAC + mic preamp) like MOTU UltraLite MK5, RME Fireface UCX II or similar. Obviously, it depends on how much you are willing to pay and their reputation and stuff like user interface, driver support etc, I have no particular recommendation.

Intersample headroom in DAC is irrelevant because room correction will significantly change the peaks of any input waveform, should it be originally loudness war content or highly dynamic audiophile stuff. The audio interface driver often provide a control panel app to let you monitor the incoming, room corrected audio data and you will know if the incoming data is clipped or not. To avoid clipping, adjust the software playback level (e.g. headroom management, level of convolver, or other volume controls internal to the playback software before negotiating with the device driver), not any subsequent volume control like the one on the interface's mixer app, or physical knobs on the interface. It is like wearing protective suit before entering radioactive area, not after, order is important.

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #4
Not your main item, but this is solvable.......
Intersample overshoots "are only a problem near full volume". If your playback application (digitally) attenuates, you have more headroom.
Solution: Play from files, with ReplayGain. Scan and tag your files, and use a ReplayGain-aware player. More securely, use a ReplayGain scanner with upsampling, which measures (to the capability of the algorithm) the true intersample peak and sets the peak tag accordingly.
foobar2000 can do that, and also limit output volume according to peak to keep you safe even when you run the player's volume slider to full.

It has been discussed in-depth at this forum too, most recently at https://hydrogenaud.io/index.php/topic,124977.0.html . In most cases, the RG gaini figure will attenuate enough. But if you set playback to be limited according to the peak tag - after scanning as above - then the problem is solved. 
Thanks for what's apparently a true solution for reducing the perception (or delivery??) of IS distortion. But to achieve this goal it does look I might will be confined to using one app for music playback, life, though it seems to be a happy compromise, so far.

For me, beyond delivering what's generally called audiophile sound quality, the single most important playback feature of any Windows player is transparent sounding pitch control. Ideally, time control would also be included. Back in the day this app (cruelly limited to Pro Tools users) was the gold standard for providing both pitch and time control. https://serato.com/pitchntime-pro Today most DAWs-include my very old version of Samplitude. https://www.soundonsound.com/reviews/magix-samplitude-pro-x

But I see here that Foobar player does have both pitch shift  and time rate playback features https://www.foobar2000.org/components/view/foo_dsp_effect  and full Replay Gain support. https://www.foobar2000.org/ I'll download it and if it's pitch and time widgets sound as clean as Samplitude's then great. But I don't know if can fully understand what Replay Gain does and how to use it to reduce/eliminate IS distortion. What's the best primer for DUMMIES to learn how?

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #5
A few points:
* There are more RG-aware players than fb2k: https://en.wikipedia.org/wiki/ReplayGain#Implementations
* Software players adjust volume in the digital domain. If you play back at 6 dB under full volume, then you are safe from inter-sample overs that are 6 dB or less. As pointed out in that thread, I have only noise music that exceed that - among lossless files, that is.
* Some DACs have already 3 dB headroom because they upsample. Check your DAC's specs.
* Turning an analog volume knob up does not change anything.

But:
Lossy files may have higher volume than digital full scale - even with no "intersample" considerations, as the audio data aren't stored that way at all. I don't know why things like these sometimes happen, but when artists do things themselves and upload ... here is an example, Devotional Songs from https://sabaziusdoom.wordpress.com/free-downloads/ . The long tracks 2 and 3 peak at > +6 dB even without true peak scan. So does the track you can download from https://soundcloud.com/riotseason/aufgehoben-anno-fauve if you are logged into a Soundcloud account. (Click More and then download.)

 

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #6
USB has no role in syncing two separate ADC and DAC, you at least need to chain them using SPDIF/AES3/ADAT for synchronization. Also, only ADAT is capable of multiple (eight) LPCM channels, SPDIF and AES3 only support stereo, and ADAT only supports up to 48kHz when using all 8 channels. Syncing can be achieved by setting one of the device as master and another one as slave, or via SRC if supported by the device's hardware. However such methods are rather cumbersome and more prone to drop out, so just get an all-in-one interface (ADC + DAC + mic preamp) like MOTU UltraLite MK5, RME Fireface UCX II or similar. Obviously, it depends on how much you are willing to pay and their reputation and stuff like user interface, driver support etc, I have no particular recommendation.

Intersample headroom in DAC is irrelevant because room correction will significantly change the peaks of any input waveform, should it be originally loudness war content or highly dynamic audiophile stuff. The audio interface driver often provide a control panel app to let you monitor the incoming, room corrected audio data and you will know if the incoming data is clipped or not. To avoid clipping, adjust the software playback level (e.g. headroom management, level of convolver, or other volume controls internal to the playback software before negotiating with the device driver), not any subsequent volume control like the one on the interface's mixer app, or physical knobs on the interface. It is like wearing protective suit before entering radioactive area, not after, order is important. 
Regarding your second paragraph, I can't quite visualize your methods for preventing IS distortion problems. But I'm hoping that IF I'm ultimately unable or choose not to pursue software based room correction that Porcus's Foobar/Replay Gain solution will solve my IS distortion issues where extant in my music sources. But tutorials for dummies would be essential.

With your first paragraph, I've never read about much less encountered anything with ADAT connectivity and don't know how it would be applicable for equalizing my system to my room. https://emastered.com/blog/what-is-adat Among other things, what I don't get is how Kal Rubinson seemed able to steer clear of all or most of the connectivity issues raised, plus without any temporary or permanent use of multiple ADC channels (beyond the one for making microphone test recordings) and yet have complete success implementing DIRAC Live 3 RC . Indeed, save for whatever mic/preamp/ADC/USB hardware he needed to run DIRAC mic test signal measurement recordings for his main speakers and subs (Revel and SVS all passively crossed I think), this Exasound e38MK2 eight-channel DAC was the only converter I read about which remained in his system, at least for MCH playback. https://www.stereophile.com/content/music-round-85-nimitra-exasound-baetis-roon

The only thing Kal did find was that use of DIRAC imposed nearly a 20db gain loss when he reviewed and used this DAC for room correction. https://www.stereophile.com/content/okto-research-dac8-pro-da-processor To make up for that loss he resorted to the rather clunky solution of installing three of these stereo preamps. https://www.stereophile.com/content/topping-pre90-line-preamplifier

But I still wish I could understand how Kal apparently achieved system/room eq if he did not have to contend the connectivity protocols you describe or what he did to circumvent those issues. https://www.audiosciencereview.com/forum/index.php?threads/audiopraise-vanitypro-review-hdmi-audio-extractor.30440/page-14

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #7
A few points:
* There are more RG-aware players than fb2k: https://en.wikipedia.org/wiki/ReplayGain#Implementations
* Software players adjust volume in the digital domain. If you play back at 6 dB under full volume, then you are safe from inter-sample overs that are 6 dB or less. As pointed out in that thread, I have only noise music that exceed that - among lossless files, that is.
* Some DACs have already 3 dB headroom because they upsample. Check your DAC's specs.
* Turning an analog volume knob up does not change anything.

But:
Lossy files may have higher volume than digital full scale - even with no "intersample" considerations, as the audio data aren't stored that way at all. I don't know why things like these sometimes happen, but when artists do things themselves and upload ... here is an example, Devotional Songs from https://sabaziusdoom.wordpress.com/free-downloads/ . The long tracks 2 and 3 peak at > +6 dB even without true peak scan. So does the track you can download from https://soundcloud.com/riotseason/aufgehoben-anno-fauve if you are logged into a Soundcloud account. (Click More and then download.)
  Foobar, VLC and JRiver implement Replay/Gain, nice! Sure about JRiver? Something about a "failed verification" here. https://en.wikipedia.org/wiki/ReplayGain

I wouldn't be surprised if Samplitude Pro X (2011 version) can be set up to do at least reduce the level below so averaged threshold, though perhaps not as elegantly and subjectively sweet sounding as I'm Replay/Gain can.
I'd have to consult the forum, though some are not keen on advising with versions that old. https://www.magix.info/us/forum/samplitude-pro-x7-music-production-software-for-audio-pros--1300091/

But as JRiver and VLC are my main go-to DVD and BD movie players will Replay/Gain provide any such IS distortion reduction for the Dolby Digital (lossy) and DTS-MA (lossless) soundtracks on my DVDs and BDs?

Still, what I don't get is that those who've discussed the cause of this distortion (distortion loving artists??) at length https://gearspace.com/board/mastering-forum/1401406-intersample-clipping-audible-17.html , maintain that what ever scheme may be implement in (consumer) DACs and/or media player software to solve the problem in a practical sense, it will not change the fact that the waveforms of those recordings are permanently clipped. However, correct me if I wrong, but does Replay/Gain work its magic largely by averaging the overall levels in the track, applying an algorithm that impressed how we hear into the playback software logic https://en.wikipedia.org/wiki/Equal-loudness_contour while then insuring that the desired peak value (however determined) is never exceeded.

However, since as the gearspace posters say those clipped peaks are there to stay, some of the track's dynamic range is traded to get a distortionless sounding recording, yes?

And this metadata would be storable in the uncompressed FLAC file that I would create to store a CD track rip? If yes, does VLC, Foobar and JRiver make it EASY for dummies to create and store that data in the FLAC file?

And just to be sure, I can add this metadata generated by Replay/Gain to any previously created FLAC file of an uncompressed CD track rip?

Or FLAC file track of a 24 bit download?
https://www.hdtracks.com/#/album/5df1427d0bee25c09bc163fd

 No doubt roon other high res proprietary players (even Youtube?) have implemented very similar strategies for reducing IS distortion.

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #8
But as JRiver and VLC are my main go-to DVD and BD movie players will Replay/Gain provide any such IS distortion reduction for the Dolby Digital (lossy) and DTS-MA (lossless) soundtracks on my DVDs and BDs?
I don't know how they work specifically, but it seems to me that VLC can boost volume above some "100 percent" mark. I have always assumed that it means it amplifies and can be forced to clip. I might be wrong.

Dolby Digital ... dunno. My "loudest AC3" measures inter-sample peaks at +1.4 dB.
Also one thing I forgot about lossies: not all those inter-samples overs are "necessary" for lossy formats.
* OK, so a lossy encoding can boost volume (it is after all not lossless, so no expectation it should encode to the same as the original - and if it can make for something that sounds good by tweaking volume, nothing says it cannot do that).
* There will every now and then be spikes as artefacts of the lossiness. Again, it is not lossless. Imagine there were no inter-sample overs before lossy compression - nobody says there need not be one after. And if overall volume isn't really changed, and those new intersample overs are type "heck, it overshoots the original signal, but you ain't gonna hear it!", then who says removing it by clipping makes it any worse? Yeah clipping is typically bad, but clipping away something that shouldn't be there, that isn't the same thing.
* Atop that, you have the inter-sample overs that were there in the lossless ... uh-oh. Hard to tell which are which. Actually some recommend that when you do lossy encoding, you do RG first and copy the tags, rather than scan the new files. I don't know whether that solves or creates any problems, to be honest.

maintain that what ever scheme may be implement in (consumer) DACs and/or media player software to solve the problem in a practical sense, it will not change the fact that the waveforms of those recordings are permanently clipped.
I didn't bother to read that.
There is a lot of irreversible clipping introduced in audio processing. (I mean, audio processing includes everything that happens from somewhere before the microphone and 'til somewhere after the loudspeakers ...) And nowadays, processing in the digital domain could very well be in 32-bit floating-point - decimating down to say CDDA, is inherently lossy (and float --> integer done in a stupid way, can create irreversible damage).
So in all this, there is some distortion introduced in the process, and some distortion that can be safeguarded against. Of course, reducing volume also reduces resolution and signal-to-noise ratio (which isn't much of a problem even with CDDA anyway, especially not for loud tracks!) - but let's say you got an unhealthy mix of claims ranging from true through "true but irrelevant" through "wrong worded but a genuine problem" through "true it could be an issue, so therefore fix it this way to avoid it" to total bogus.

Let's just say, volume reduction solves some problems - those which are due to too loud volume, and that includes those which just have an "intersample peak problem" in the final digital waveform.
Of course if things have been back and forth and gotten irreversible damage - including intentional loudness jaywalking - then nothing says you can get those pieces of the signal back.


However, correct me if I wrong, but does Replay/Gain work its magic largely by averaging the overall levels in the track,
Well, kind of. There are 2x2=4 RG tags. One distinction is between "track" and "album"; the latter works by grouping together all the tracks, presuming you want one volume setting for the full album. But in principle, that isn't different except in what it does scan. So let's for the sake of the discussion just assume your "album" is a 1-track single, and those will coincide.
Then there are two tags. Gain and peak. Gain suggests how to attenuate (or sometimes boost) it to get everything sound closer to "equally loud". Peak is ... peak. Measured the way it is measured when scanning, good or bad.
Both are just numbers.
I set foobar2000 to
* attenuate everything that has no RG tags, by 9 dB.
* use gain but limit according to peak. That means, fb2k will try to use the gain figure, but if that is so loud it runs into clipping, it just bumps down the volume until clipping is gone.

(Why that "attenuate" and merely "sometimes" boost? Because it sets a target loudness, and that is set for "lower than average" to ensure that most music is indeed attenuated. Why? To prevent clipping! Or if you like, to prevent the "peak" figure from kicking in too often.)


And this metadata would be storable in the uncompressed FLAC file that I would create to store a CD track rip? If yes, does VLC, Foobar and JRiver make it EASY for dummies to create and store that data in the FLAC file?

And just to be sure, I can add this metadata generated by Replay/Gain to any previously created FLAC file of an uncompressed CD track rip?
Yes. It doesn't change the encoded data. It is just a tag that says "hey player, turn down volume by 8 dB when you play this album". Delete the tag, and you are back to fresh.
It doesn't always look like that in your player, because some players "hide it from your user-taggable view": they presume that you might want to add a Comment tag, but you don't want to manually edit the RG figures. But others just let you tweak it manually. Into "hey player, turn down volume by 8.76 dB when you play this album", or whatever.

Well, a warning: Do not use functionality called e.g. "ReplayGain (apply)". That will - typically - change the volume of your encoded audio data.
So as long as you can: stick to tags.

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #9
Regarding your second paragraph, I can't quite visualize your methods for preventing IS distortion problems. But I'm hoping that IF I'm ultimately unable or choose not to pursue software based room correction that Porcus's Foobar/Replay Gain solution will solve my IS distortion issues where extant in my music sources. But tutorials for dummies would be essential.
ReplayGain data is collected by scanning the source materials without taking room correction into account.

Because you are doing room correction and room correction is basically a customized EQ applied to the source. After room correction the waveform will change and ReplayGain peak values based on the source cannot precisely reflect the peak values after room correction.

If you still want to do it in the RG way you will need to store the room corrected audio data into a separate set of audio files using a lossless floating point PCM format, then perform RG scan on these floating point files to find the actual, room corrected peaks.

Hopefully you already understand the differences between sample over and intersample over. Headroom in the DAC can only mitigate intersample over, not sample over, and room correction can introduce sample over. Regarding the Gearspace thread you pointed to, the OP of the thread (Ian) in fact confused about different types of over and made a clarification:
https://youtu.be/uHA2c9NrBPk
If you are doing room correction on the fly, then you are sending the room corrected audio data to the audio driver, and even my cheapo Sound Blaster has a realtime meter to monitor the room corrected input level to the device driver. Basically, you just lower the software playback level to the device driver so that the meter peaks at several dBs below full scale and the job of preventing intersample over is done.


Quote
With your first paragraph, I've never read about much less encountered anything with ADAT connectivity and don't know how it would be applicable for equalizing my system to my room. https://emastered.com/blog/what-is-adat Among other things, what I don't get is how Kal Rubinson seemed able to steer clear of all or most of the connectivity issues raised, plus without any temporary or permanent use of multiple ADC channels (beyond the one for making microphone test recordings) and yet have complete success implementing DIRAC Live 3 RC . Indeed, save for whatever mic/preamp/ADC/USB hardware he needed to run DIRAC mic test signal measurement recordings for his main speakers and subs (Revel and SVS all passively crossed I think), this Exasound e38MK2 eight-channel DAC was the only converter I read about which remained in his system, at least for MCH playback. https://www.stereophile.com/content/music-round-85-nimitra-exasound-baetis-roon

The only thing Kal did find was that use of DIRAC imposed nearly a 20db gain loss when he reviewed and used this DAC for room correction. https://www.stereophile.com/content/okto-research-dac8-pro-da-processor To make up for that loss he resorted to the rather clunky solution of installing three of these stereo preamps. https://www.stereophile.com/content/topping-pre90-line-preamplifier

But I still wish I could understand how Kal apparently achieved system/room eq if he did not have to contend the connectivity protocols you describe or what he did to circumvent those issues. https://www.audiosciencereview.com/forum/index.php?threads/audiopraise-vanitypro-review-hdmi-audio-extractor.30440/page-14
Synchronization is required only if you want to operate two independent devices simultaneously with one clock source. It is not required if you only use the ADC to capture a room correction profile and use the profile during playback. I just wanted to point out USB has no role on syncing two independent devices.

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #10
ReplayGain data is collected by scanning the source materials without taking room correction into account.

Because you are doing room correction and room correction is basically a customized EQ applied to the source. After room correction the waveform will change and ReplayGain peak values based on the source cannot precisely reflect the peak values after room correction.

If you still want to do it in the RG way you will need to store the room corrected audio data into a separate set of audio files using a lossless floating point PCM format, then perform RG scan on these floating point files to find the actual, room corrected peaks.

If you are doing room correction on the fly, then you are sending the room corrected audio data to the audio driver, and even my cheapo Sound Blaster has a realtime meter to monitor the room corrected input level to the device driver. Basically, you just lower the software playback level to the device driver so that the meter peaks at several dBs below full scale and the job of preventing intersample over is done.
Quote
I think I get most of this though I need a good definition of what's meant by floating point in such as audio data file.
 

The only thing Kal did find was that use of DIRAC imposed nearly a 20db gain loss when he reviewed and used this DAC for room correction. https://www.stereophile.com/content/okto-research-dac8-pro-da-processor To make up for that loss he resorted to the rather clunky solution of installing three of these stereo preamps. https://www.stereophile.com/content/topping-pre90-line-preamplifier

But I still wish I could understand how Kal apparently achieved system/room eq if he did not have to contend the connectivity protocols you describe or what he did to circumvent those issues. https://www.audiosciencereview.com/forum/index.php?threads/audiopraise-vanitypro-review-hdmi-audio-extractor.30440/page-14
Synchronization is required only if you want to operate two independent devices simultaneously with one clock source. It is not required if you only use the ADC to capture a room correction profile and use the profile during playback. I just wanted to point out USB has no role on syncing two independent devices.
  Okay, that's pretty clear. Actually, it looks like Kal now uses Merging HAPI, which I think includes DACs and ADCs, so then no sync issues, yes? https://www.audiosciencereview.com/forum/index.php?threads/room-correction-hardware-software-issues-options-please-advise.49900/#post-1789735

However, it just occurred to me that as so many of my favorite recordings were made long before the loudness wars
https://www.hdtracks.com/#/album/5df1427d0bee25c09bc163fd
https://www.hdtracks.com/#/album/6008a730f08b3e5a17bf3918
https://www.hdtracks.com/#/album/5de17d703f6dd7ed93fd9bdc
https://www.hdtracks.com/#/album/5def9fdcb45f07686f019c26
https://en.wikipedia.org/wiki/Dreams_Are_Nuthin%27_More_Than_Wishes
https://www.hdtracks.com/#/album/5e0336d482679ee0f64706b2
https://en.wikipedia.org/wiki/Herb_Alpert_Presents_Sergio_Mendes_%26_Brasil_%2766
https://en.wikipedia.org/wiki/Equinox_(S%C3%A9rgio_Mendes_album)
https://en.wikipedia.org/wiki/Look_Around_(S%C3%A9rgio_Mendes_album)
https://en.wikipedia.org/wiki/The_Best_of_Sade

………………………………….that as long as they were not later remastered by these same kinds of knuckleheads shouldn't those recordings be free of audible IS distortion?

 

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #11
Okay, that's pretty clear. Actually, it looks like Kal now uses Merging HAPI, which I think includes DACs and ADCs, so then no sync issues, yes? https://www.audiosciencereview.com/forum/index.php?threads/room-correction-hardware-software-issues-options-please-advise.49900/#post-1789735
Right, all I/O channels share a single internal clock so they are all in sync.

Quote
However, it just occurred to me that as so many of my favorite recordings were made long before the loudness wars
https://www.hdtracks.com/#/album/5df1427d0bee25c09bc163fd
https://www.hdtracks.com/#/album/6008a730f08b3e5a17bf3918
https://www.hdtracks.com/#/album/5de17d703f6dd7ed93fd9bdc
https://www.hdtracks.com/#/album/5def9fdcb45f07686f019c26
https://en.wikipedia.org/wiki/Dreams_Are_Nuthin%27_More_Than_Wishes
https://www.hdtracks.com/#/album/5e0336d482679ee0f64706b2
https://en.wikipedia.org/wiki/Herb_Alpert_Presents_Sergio_Mendes_%26_Brasil_%2766
https://en.wikipedia.org/wiki/Equinox_(S%C3%A9rgio_Mendes_album)
https://en.wikipedia.org/wiki/Look_Around_(S%C3%A9rgio_Mendes_album)
https://en.wikipedia.org/wiki/The_Best_of_Sade

………………………………….that as long as they were not later remastered by these same kinds of knuckleheads shouldn't those recordings be free of audible IS distortion?
I don't have your files so can't answer it for you. Just want to give an example that some apparently non-loudness war contents can contain high intersample overs:
https://www.audiosciencereview.com/forum/index.php?threads/lets-develop-an-asr-inter-sample-test-procedure-for-dacs.49050/post-1763519
Read the replies as well, some people posted their ABX results.

Re: Room Correction Hardware/Software Issues, Options. Please Advise.

Reply #12
I went there and tried my best but grasped maybe 40% of the discussion. Maybe if I started reading that thread from page 1?? Otherwise, I don't think I will get a better practical understanding of the problem until I attempt to use Replay/Gain in Foobar or JRiver-and/or other solutions you guys may have-to reduce IS distortion in special content.
Also, what might help me better understand is if there's some app in JRiver or Foobar that can draw spectrograms like this. https://www.audiosciencereview.com/forum/index.php?attachments/t_t480s-png.324418/ That looks like something Izotope Rx draws. https://www.izotope.com/en/products/rx.html