• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

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

Balle Clorin

Major Contributor
Joined
Dec 26, 2017
Messages
1,430
Likes
1,295
I got a second hand Trinnov Amethyst and use it with Roon for streaming Tidal and hard disk tracks, and CD and Vinyl.
The only problem is the price, but I got it for 50%. Now worries , fantastic support if I wonder about something. A bit of a learning curve to utilise all settings possible. The unit is now 10 years, maybe something new is coming, the DACs can be upgraded and replaced by the owner to 2023 status.(disconnect cables and swap a card)

Yes, it makes a difference to the sound. My best purchase ever…
 

dualazmak

Major Contributor
Forum Donor
Joined
Feb 29, 2020
Messages
2,938
Likes
3,168
Location
Ichihara City, Chiba Prefecture, Japan
The other good news is that most of dualazmak's sources are CDs, and who apparently gets wonderful results by PCM upsampling CD tracl rips to 88.2kHz. Btw, do you rip to uncompressed WAV or FLAC. Which file format, and why? If you save to FLAC files do you run check sums to look for bit errors in the rip?

For the details of my digital music library organization, you would please refer to my post here. As I wrote there, I have been ripping CD into bit-perfect 44.1 kHz 16 bit AIFF format (bit-perfect non-compressed intact PCM, like WAV format).

I was using Apple iTunes on Windows PC when I started ripping CDs into AIFF format with very much flexible tag info capabilities using iTunes. Soon after, I have abandoned using iTunes replacing it with JRiver MC for ripping (continuing into AIFF) and playing. Then, soon again I moved to dBpoweramp CD Ripper which was/is very accurate having checksum function and really nice web tag search ability, still ripping into AIFF.

Nowadays, I frequently use JRiver MC for tag editing since JRiver memorizes all the tag info (e.g. composers, artists, etc.) of the music tracks so far accumulated in its digital library and easy to be selected from the pull-down list of existing tag info.

Whatever side of the issue you may find yourself on, since it will be months before I would get close to making this test myself, if you happen the Steely Dan album Siau referenced
https://en.wikipedia.org/wiki/Two_Against_Nature (or your local public library may have a copy), please
rip the track "Gaslighting Abbie".

Sorry, I too do not have that CD and have no convenient free access to the copy of that CD.
I am looking forward to hearing results of your own analysis/measurement on the specific track of that CD to be hopefully kindly shared with us very soon!
 

ZolaIII

Major Contributor
Joined
Jul 28, 2019
Messages
4,328
Likes
2,565
I personally think that DSD upsampling will not give you much audible benefit, but there are plenty of people who do. It is possible to use software to convert PCM to DSD and upsample it in real time, the only software I am aware of that can do this is HQPlayer. If you do use HQPlayer, you need to be aware of the following caveats:

- HQPlayer is expensive! USD$326.
- Performing 6-8 channels of conversion + convolution is EXTREMELY CPU intensive. It is not uncommon for people to use dedicated PC's for HQPlayer, and these are typically fire breathing watercooled monsters with high end graphics cards (HQP takes advantage of nVidia CUDA cores) typical of gaming setups.

If you decide to use HQP, you can use either the embedded version (where it lives on its own dedicated PC) or the player version (where you use a third party playback program like Roon on the same computer). Again, there are people who believe that separating the convolution/upsampling function from playback into two PC's results in better sound. I am not one of them.

Don't forget that the thermal cost of using HQP is significant and you will need a strategy to deal with that. If you decide to air cool your PC, your CPU fans will be running all the time and it will raise your room's noise floor unless you move your PC to another room and run a long network cable to send output from the PC into your Merging via Ravenna. Or you could consider watercooling, which does reduce noise, but increases the complexity of your build.

Now, I know for a fact that my Merging NADAC has a lower SINAD when fed a DSD signal (-118dB) compared to 44.1kHz PCM (-110dB). I have measured it with an E1DA. However, upsampling PCM from 44.1kHz to 192kHz also brings the SINAD down to about -116dB. The CPU cost for upsampling PCM 44.1kHz --> PCM 192kHz is minimal, my CPU typically sits at 8% usage when I am doing this. OTOH, upsampling PCM 44.1kHz --> DSD128 results in CPU usage of 95%, and I sometimes get audible signal dropout. The benefit is a 2dB improvement in SINAD. I can tell you that I can not hear a difference between PCM 44.1kHz and PCM 192kHz, and I definitely can not hear a difference with DSD. The reason I upsample to 192kHz is because it is a "free upgrade" in SINAD and it only costs me a tiny bit extra CPU usage and raises my electricity bill by a tiny amount.

I spoke earlier in this thread about our own biases, and this is one of mine. IN MY OPINION, DSD upsampling is not needed. If it came with a low CPU cost, I would do it and grab the extra -2dB improvement in SINAD even though I know perfectly well it is not audible. What you are really doing is expanding a lot of effort into pushing an inaudible number into an even more inaudible number. There is plenty of low hanging fruit to pick before you even do that. I can tell you for a fact that creating better convolution filters through improvement in my understanding of room acoustics, measurement methods, knowing what to correct/what not to correct, and better usage of my software (Acourate) results in massive objective and subjective improvements to the sound.

However, there are people who want "the ultimate" and build two PC's, upsample to DSD, electrically isolate all network connections by using optical, use atomic clocks, audiophile grade switches, and the like ... here is an example of such a system. I applaud his effort, determination, and attention to detail. In fact I have to say that I admire him, even if I disagree with half the things he is doing. You can decide if you want to go down that path, but I doubt that many people on ASR would think that going to such lengths is fruitful.
There is only one written implementation for PCM to DSD conversation and you do it after processing. SINAD diference between DSD 128 and 24 bit PCM in heatable range is not even academic. That only ever written implementation is semi SMP 4 integer optimised and needs OoO quad core CPU. It's single most CPU intensive tast there is when it comes to audio (not that it's a much of a problem on modern PC).
You can do it in foobar2000 with plugin or something like JRiver.
PCM to DSD can be only beneficial in rare cases of early ESS 32 gen where separate line for it is better autended and even then with output level (V) cost. In any other case you don't get anything other than inconvenience and penalties.
Peple tend not to understand PDM's advantage in transport is no need for a complex bus and syncing (no introduced jitter, no bus or sync clocks [oscillation crystals] in design and list ends there so in limited application use cases it will always have it's design place. Regarding it as musical format or for general storage you absolutely don't want DSD!

@dped90 to cut it short do processing on FP PC and get as much physical chenels you need and to desired performance level with inputs you desire and on to balanced/unbalanced output path. Learn to use REW and preferably with UMIK-1. Take your time and patience and you will get there.
If you want to go with self operating integrated solution inspire to their limitations (processing accuracy, number of taps/PEQ's per bank/chenel, no loudness, filter quality...) and price for 4 to 8 chenel solutions take a look at MiniDSP Flex/Flex HT offering. If you are into "pro audio" take a look at Motu audio interfaces and or even power amplifiers with usable deacent and adequate performing embedded DSP (ADC-DSP-DAC) in mid price category. If you are OK with desktop PC (not mini or burbon) and 6 chenels are enough and you prefer unbalanced and don't really need a good ADC even sometimes like Creative AE-5/5 Plus will do just fine.
From AVR's check Denon's offering of course they have their own set of limitations employed.
Regarding semi auto corrections they will always do more or less good job but never as you in the end (and for a lots of reasons including double individual ones [room and you]).
Regarding software JRiver and on Windows only (WDM driver) for not too complex number of chenels (as remaining is pain in the... and you can not rename them) setups is probably best most complete/comprehensive offering there is (to my better knowledge and to this point in existence).
It depends how you intend to get there and what do you need after all but be sure you won't really get there without considerable self employment, learning curve and time spent.
 
OP
D

dped90

Active Member
Joined
Feb 12, 2022
Messages
197
Likes
24
Location
NY
Thanks to everyone here who have made it crystal clear that PCM to DSD conversion for upsampling offers NO benefits and only creates overheated, fan noise crazy PCs and higher electric bills. And what's nice is that I can opt for the cheaper DA8 card for the Hapi, as the DA8P card with DSD is apparently only for those playing SACDs or rips of same.

OTOH, "basic" PCM upsampling of ripped CD tracks to 88.2kHz apparently provides substantially audible improvement in sound quality and with penalties of any kind.
But my question regarding this is can such upsampling of FLAC (or WAV) files of CD track rips be done by the Merging Hapi DA8 card set up to do room EQ for my six channel speaker/subwoofer system using Windows players like JRiver or Foobar 2000.

Furthermore, can it do so with Replay Gain running to normalize (?) loudness levels?
https://hydrogenaud.io/index.php/topic,124977.0.html https://wiki.hydrogenaud.io/index.php?title=ReplayGain_specification#Clipping_prevention
 
Last edited:

dualazmak

Major Contributor
Forum Donor
Joined
Feb 29, 2020
Messages
2,938
Likes
3,168
Location
Ichihara City, Chiba Prefecture, Japan
OTOH, "basic" PCM upsampling of ripped CD tracks to 88.2kHz apparently provides substantially audible improvement in sound quality and with penalties of any kind.
Please note that I definitely never said that "audible improvement in sound quality" given by upsampling to 88.2 kHz.
I said that 88.2 kHz DSP processing is just fine enough at least in my multichannel DSP audio rig and for my ears+brain (ref. my post #532 on my project thread).

But my question regarding this is can such upsampling of FLAC (or WAV) files of CD track rips be done by the Merging Hapi DA8 card set up to do room EQ for my six channel speaker/subwoofer system using Windows players like JRiver or Foobar 2000.
I still do not fully understand what would be your point. Why are you asking "Merging Hapi" to do upsampling??

I upsample/downsample all the tracks into 88.2 kHz PCM by JRiver's on-the-fly (while playing music) format conversion;
Tools - Options - DSP & output format - DSP Studio - Output Encoding="None", Sample rate: Output (right click): Set all to 88,200 Hz

As for JRiver's "Memory Play", my setting is;
Tools - Options - Memory playback - Load full file (not encoded) into memory

Furthermore, can it do so with Replay Gain running to normalize (?) loudness levels?
For master volume control, JRiver's internal volume can take care of it.

As for relative gain controls between the DSP-ed multiple channels, the system-wide DSP software ("EKIO" in my case) takes care of them.

As for gain/loudness matching between DSP-ed ASIO channels and Windows default WDM sound device, you can do it by using system-wide audio routing tools, in my case VB-Audio Matrix (ref. here #858 on my thread).


Am I responding suitably/properly to your inquiries?
 
Last edited:

Keith_W

Major Contributor
Joined
Jun 26, 2016
Messages
2,885
Likes
6,668
Location
Melbourne, Australia
OTOH, "basic" PCM upsampling of ripped CD tracks to 88.2kHz apparently provides substantially audible improvement in sound quality and with penalties of any kind.
But my question regarding this is can such upsampling of FLAC (or WAV) files of CD track rips be done by the Merging Hapi DA8 card set up to do room EQ for my six channel speaker/subwoofer system using Windows players like JRiver or Foobar 2000.

Upsampling from 44.1kHz can provide a measurable benefit, depending on your DAC. I wouldn't say "substantially audible improvement", in fact I did say that I can measure it but I can't hear it.

The upsampling is usually done by software. I don't know if the Merging Hapi can upsample by itself. Some DAC's are able to, some can't. It's another DAC specific feature. Virtually all convolvers I am aware of are able to upsample. To keep things simple, for now you do not need to consider a third party convolver. The built in convolvers in JRiver, Foobar, and Roon are "good enough" but lack some niceties in third party convolvers.


If you software supports it, then yes.
 

soerenssen

Addicted to Fun and Learning
Joined
Jul 25, 2023
Messages
558
Likes
125
I doubt if there are DACs that are audibly superior to the Okto.
Do you think it makes any sense to pair the Okto DAC8 pro with a well-measuring budget Purifi or NCx500 based power amp from Audiophonics/Nord/Apollon for KEF Reference 3 Meta mains? Alternatively I'm considering the Flex, Motu Mk5 and DM7, with Dirac/DLBC running on a computer, but other than features (like the very convenient AES/USB mode), what audible difference should I expect?
 

Kal Rubinson

Master Contributor
Industry Insider
Forum Donor
Joined
Mar 23, 2016
Messages
5,370
Likes
10,018
Location
NYC
Do you think it makes any sense to pair the Okto DAC8 pro with a well-measuring budget Purifi or NCx500 based power amp from Audiophonics/Nord/Apollon for KEF Reference 3 Meta mains?
Sure, it does.
Alternatively I'm considering the Flex, Motu Mk5 and DM7, with Dirac/DLBC running on a computer, but other than features (like the very convenient AES/USB mode), what audible difference should I expect?
The DM7 and the Okto are audibly equivalent as DACs. The Okto has additional features which, for some of us, justify its higher price.

I have 4 mch DACs and this is my order of preference for general use: Merging Hapi II, exaSound s88, Okto Dac8 Pro, Topping DM7. (Yes, that is also the order of price from high to low.) For some specific situations, the order will change based on the specific unique features of each.
  • Hapi II: 16 channels, Ravenna/AES67
  • exaSound: Wired and wireless input
  • Okto: AES/EBU input
  • DM7: ???
I am not familiar with Flex or Motu Mk5.
 

soerenssen

Addicted to Fun and Learning
Joined
Jul 25, 2023
Messages
558
Likes
125
Okto: AES/EBU input
DM7: ???
Thank you for you help!
AES/EBU would be the only reason in my case why I would go with the Okto and not the Topping, so I'm trying to figure out how I could connect the WiiM optical to USB on the PC host, apply Dirac and route the output to the DM7's analogue outputs via USB, while keeping them in sync. If it's not feasible, then I'll either stream directly from the PC or just get the Okto (or maybe the Mk5) for this feature alone.
 

Keith_W

Major Contributor
Joined
Jun 26, 2016
Messages
2,885
Likes
6,668
Location
Melbourne, Australia
Thank you for you help!
AES/EBU would be the only reason in my case why I would go with the Okto and not the Topping, so I'm trying to figure out how I could connect the WiiM optical to USB on the PC host, apply Dirac and route the output to the DM7's analogue outputs via USB, while keeping them in sync. If it's not feasible, then I'll either stream directly from the PC or just get the Okto (or maybe the Mk5) for this feature alone.

If you want to do that, you will need a pro audio interface like the RME Fireface UC, or the Motu Mk.5, or even the Merging Hapi. This allows you to route channels through the interface into the PC and back out again. I may be wrong, but I don't think the Okto DAC8 has this feature. This is how you would set it up:

Wiim --> (via optical) --> Fireface UC ADAT input
Fireface UC --> (via USB) --> PC
PC: set up to receive input from ADAT input on Fireface UC, and output to channels 1-8 on Fireface UC ***
PC --> (via USB) --> Fireface UC

*** this step is more complicated than you might think, because it depends on your player software / convolver. Some software, like HQPlayer Embedded, makes it easy - it's "set and forget" and no need to reconfigure every time you turn it on. Others, like JRiver, requires you to "Open ASIO session" and set it to listen for ASIO input via the ADAT port. Some other software, like Roon, can't do it at all. I am not sure how Hang Loose Convolver handles this, so i'll tag @mitchco and see if it is possible.
 

soerenssen

Addicted to Fun and Learning
Joined
Jul 25, 2023
Messages
558
Likes
125
If you want to do that, you will need a pro audio interface like the RME Fireface UC, or the Motu Mk.5, or even the Merging Hapi. This allows you to route channels through the interface into the PC and back out again. I may be wrong, but I don't think the Okto DAC8 has this feature. This is how you would set it up:

Wiim --> (via optical) --> Fireface UC ADAT input
Fireface UC --> (via USB) --> PC
PC: set up to receive input from ADAT input on Fireface UC, and output to channels 1-8 on Fireface UC ***
PC --> (via USB) --> Fireface UC

*** this step is more complicated than you might think, because it depends on your player software / convolver. Some software, like HQPlayer Embedded, makes it easy - it's "set and forget" and no need to reconfigure every time you turn it on. Others, like JRiver, requires you to "Open ASIO session" and set it to listen for ASIO input via the ADAT port. Some other software, like Roon, can't do it at all. I am not sure how Hang Loose Convolver handles this, so i'll tag @mitchco and see if it is possible.
Okto has AES/EBU input, so I could connect the Wiim's coax output directly to the Okto and it supports bi-directional USB audio.
The Motu Mk5 has both optical and SPDIF coax inputs and can do the same as far as I know.
 

mitchco

Addicted to Fun and Learning
Audio Company
Forum Donor
Joined
May 24, 2016
Messages
646
Likes
2,423
Some other software, like Roon, can't do it at all. I am not sure how Hang Loose Convolver handles this, so i'll tag @mitchco and see if it is possible.
Okto has AES/EBU input, so I could connect the Wiim's coax output directly to the Okto and it supports bi-directional USB audio.
The Motu Mk5 has both optical and SPDIF coax inputs and can do the same as far as I know.
Yes, it is set and forget. In either case, of the Okto or Mk5, one can select the ASIO driver and see both inputs and outputs. Here is an example of the mk5 I/O in Hang Loose Host app:

1702244247509.png


Just select whichever inputs and output as required.

And it does not have to be just Hang Loose Convolver, one can add Dirac's processor as well:

index.php


Or uBACCH or Auro3D or whatever plugin you desire and wire it up anyway you like.
 

dualazmak

Major Contributor
Forum Donor
Joined
Feb 29, 2020
Messages
2,938
Likes
3,168
Location
Ichihara City, Chiba Prefecture, Japan
Okto has AES/EBU input, so I could connect the Wiim's coax output directly to the Okto and it supports bi-directional USB audio.
The Motu Mk5 has both optical and SPDIF coax inputs and can do the same as far as I know.
Yes, just for our reference, and for sure, we can find/confirm it on Okto Research web site;
https://www.oktoresearch.com/dac8pro.htm
WS00006640.JPG
 
Last edited:
Top Bottom