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HOLO Audio Spring DAC - R2R DSD512


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If I understand it correctly it is the manufacturers (Auralic) that need to find the fix rather the DAC manufacturers (Holo Audio Spring)? The Auralic Aries plays DSD64/DSD128 (Default is DoP) with no problem; no pop or click sound. The moment it just about to start playing DSD256 (Default is Native) a loud pop sound happened. If one pause and start playing again, a loud pop can be heard.

 

Not sure how all this works but in my DIY DAC, I have a USB board from JLSounds which supports Linux which is common OS for many boxes like Auralic and uRendu.

 

At first, I had a earlier JLSounds USB board that did not support native DSD (non-DoP) so I had to use DoP. With that board revision, I had loud pops so it was a problem with their DoP implementation. From what I know, it has to do with muting the signal while the DAC (USB board) makes the switch from PCM to DSD and/or vise versa.

 

About a year or so ago, JLSounds updated their board (firmware) to include native DSD as well as better support for Linux. As part of that process, JLSounds had to submit their firmware (?) to the Linux guys at GitHub. You can see that Holo DAC was added on August 21.

 

Currently, I can play native DSD up to DSD256 without any issues. Well not exactly, I still have very slight clicks between tracks on a very limited number of albums.

 

You still need to make sure that the version of Linux is being used by Auralic, uRendu, etc. includes this update. Also, the DAC (USB board) has the proper firmware.

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Thanks, it looks like the manufacturer (Auralic) may need to look into it. I've contacted them regarding on this issue. A look at GitHub, I saw Holo Spring Audio was added quite recent.

 

A temporary way to make DSD256 playable on Auralic Aries, I make a playlist on all the DSD256 tracks or put them into queue mode, since loud pop will start on the very first track that is being played, lower down the volume or mute it, after it start playing, increase the volume to the listening level. There will be no pop sound until it finishes the playlist or queue.

 

Please be wary that the loud pop can damage the speakers and amplifier if the level is set too high. Do it with caution until a full fix from manufacturers.

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  • 2 weeks later...

@ted_b

Any progress being made here?

It would be great if the community can come up with a solution to this frustrating problem that appears to apply, to some degree, to most DACs that can do native DSD with the microRendu.

 

DoP does not exhibit what direct does. Singxer and others, via uRendu (DoP) is silent. It is not uRendu fault per se; it seems like some dacs (Holo, iFi, others) have a real issue in Linux going direct without a loud POP at first sight of DSD. Once there, though, all is well (until stop). I manage it through volume but it is a PITA!!

 

Note: Jesus and company are trying at their end. I just now got off a teamview with Jesus where we tried some alpha stuff to address this issue. There is still work to do.

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No, I continue to do two things:

1) manage playlists to reduce or eliminate any PCM to DSD switching

2) manage the remote volume buttons REALLY well :)

 

It would be great to get rid of this issue, but the microRendu-as-NAA is better enough than Windows-as-NAA (no thumps) to make it worthwhile.

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Some people preferred the basic level Spring in NOS mode compared to other reference DACs and the inverse in OS mode, and most agree the Spring sounds best in NOS mode. Do you know if the Spring has an analog low-pass (reconstruction filter), and what type of filter it is (curious if it is a sharp filter between 20 and 22.05 with ripples or something more gentle that would work well if upsampling in the computer)?

I don't care much about DSD, and would probably not use OS. Many may think the same. In that case wouldn't it be better to transfer the cost of something that a lot of people may not use to a better USB implementation? Or remove the USB completely and sell it as a separate module...

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I agree, the microRendu is too good to miss out in. On the other hand, I often enjoy total random playback with the software surprising me with the music selection, something else I do not want to miss out in. In this case pop management is not practical.

I know smart folks look at this issue from time to time and I am hopeful that one day someone will figure out a solution.

 

No, I continue to do two things:

1) manage playlists to reduce or eliminate any PCM to DSD switching

2) manage the remote volume buttons REALLY well :)

 

It would be great to get rid of this issue, but the microRendu-as-NAA is better enough than Windows-as-NAA (no thumps) to make it worthwhile.

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Okay guys, please correct me if I'm wrong on these points:

 

1. No XMOS device will handle DSD512.

 

2. Only Amanero devices will handle DSD512.

 

3. Only way to get DSD512 out of your PC is via USB.

 

4. Bug in the Linux driver for Amanero keeps DSD512 from working in Linux systems (i.e., MicroRendu).

 

5. The Pink Faun is a PCM device only and will never be able to handle native DSD.

 

6. There is nothing at all that will work with DSD1024.

 

7. The only way to defeat the SRC stage in a DS type DAC is to meet the following requirements:

A: Use an I2S interface,

B: Send native DSD through the I2S interface,

C: DAC implementation is programmed to allow skipping of the SRC.

 

 

Sent from my iPad using Computer Audiophile

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I agree, the microRendu is too good to miss out in. On the other hand, I often enjoy total random playback with the software surprising me with the music selection, something else I do not want to miss out in. In this case pop management is not practical.

I know smart folks look at this issue from time to time and I am hopeful that one day someone will figure out a solution.

 

I too really enjoy random play John, from my entire library, both pcm and dsd. I keep the vol low enough that the popping "isn't bad".

 

For DSD to be done correctly, DAC must mute it's output for first and last 50 milliseconds of DSD data. Otherwise you are going to have clicks'n'pops.

 

Granted I often screw things up lol, if I recall correctly, wasn't it said before that it was a Linux problem that caused the pops, not the dac? Seems this is an area that there continues to be debate/deflection between whether it's a software and/or hardware problem. This is such a pain, I wish this could be something that could be fixed with either a software or firmware update...

Ryzen 7 2700 PC Server, NUC7CJYH w. 4G Apacer RAM as Renderer/LPS 1.2 - IsoRegen/LPS-1/.2 - Singxer SU-1/LPS1.2 - Holo Spring Level 3 DAC - LTA MicroZOTL MZ2 - Modwright KWA 150 Signature Amp - Tidal Audio Piano's.  

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For DSD to be done correctly, DAC must mute it's output for first and last 50 milliseconds of DSD data. Otherwise you are going to have clicks'n'pops.

 

OK, I understand that, but why does Holo (and others) Windows ASIO driver work fine, with no pops, but Linux driver pops? Seems not dac-based but OS based (or at least driver based).

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Okay guys, please correct me if I'm wrong on these points:

 

1. No XMOS device will handle DSD512.

 

2. Only Amanero devices will handle DSD512.

 

3. Only way to get DSD512 out of your PC is via USB.

 

4. Bug in the Linux driver for Amanero keeps DSD512 from working in Linux systems (i.e., MicroRendu).

 

5. The Pink Faun is a PCM device only and will never be able to handle native DSD.

 

6. There is nothing at all that will work with DSD1024.

 

7. The only way to defeat the SRC stage in a DS type DAC is to meet the following requirements:

A: Use an I2S interface,

B: Send native DSD through the I2S interface,

C: DAC implementation is programmed to allow skipping of the SRC.

 

 

Sent from my iPad using Computer Audiophile

 

Please move this to the proper thread, as this is the Holo dac thread and has nothing to do with Amanero, Pink Faun, etc. The Holo dac is indeed XMOS-based and I'm getting DSD512 (as reported dozens and dozens of times!) from both Linux (microRendu) and Windows (ASIO driver). I run NOS, so SRC is also not an issue. Your list is both inaccurate and largely irrelevant to this thread.

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Please move this to the proper thread. The Holo dac is XMOS-based and I'm getting DSD512 (as reported dozens and dozens of times!) from both Linux (microRendu) and Windows (ASIO driver). Your list is both inaccurate and irrelevant to this thread.

 

Touchy....

 

May I ask what version of the XMOS driver you have that supports DSD512 -- which I assume is native (not DoP)?

 

 

Sent from my iPad using Computer Audiophile

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GUTB, you have a CA history of not reading or understanding, just passing out false accusations. It started with this Holo dac, and my review...(only does PCM, converts DSD to PCM, yada yada ) so yes, a bit touchy cuz you troll too often for my tastes. Ask Vinnie or any of the other good folks here!

 

Read the damn review; all your info (driver numbers, DSD512 stories, etc) are there. It's been out now for several months! And discuss Amanero, Pink Faun an other non-Holo issues elsewhere. PLEASE!

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Please move this to the proper thread, as this is the Holo dac thread and has nothing to do with Amanero, Pink Faun, etc. The Holo dac is indeed XMOS-based and I'm getting DSD512 (as reported dozens and dozens of times!) from both Linux (microRendu) and Windows (ASIO driver). I run NOS, so SRC is also not an issue. Your list is both inaccurate and largely irrelevant to this thread.

 

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Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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For DSD to be done correctly, DAC must mute it's output for first and last 50 milliseconds of DSD data. Otherwise you are going to have clicks'n'pops.

 

Okay, it seemed this click issue is a DAC side issue which manufacturers have to take care and apply the correct mute function.

 

What about switching between PCM to DSD or vice-versa? When switching occurs there's brief period when the OS (Linux or Windows) or and playback program may generates all '0' data, in the case of PCM this is translated to 0V output which is fine when the DAC is in PCM mode. In a circumstances if it happens that DAC is in DSD mode, it will generates a maximum negative voltage resulting in a loud pop sound!

 

I can live with clicks but not loud pop, it will definitely damage your amps and speakers, it like sending a DC voltage to fry them!!!

 

FYI: For DSD, all '1' is maximum positive voltage, a combination of '1' and '0' or '0' and '1' is 0V output and all '0' is maximum negative output voltage. This is based on PDM (Pulse Density Modulation) use in DSD.

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OK, I understand that, but why does Holo (and others) Windows ASIO driver work fine, with no pops, but Linux driver pops? Seems not dac-based but OS based (or at least driver based).

 

Do you think this loud pop sound is caused by Linux OS or and playback program under Linux environment? I've something similar but in Windows using Roon. In DSD playback I got loud pop when I 'pause' and then start 'Play' or skip to next track. Roon managed to fix this issue. What they told me is when during 'pause' and start playing, there's a brief moment the program will generate so call PCM silence; all '0'. Unfortunately some DSD DACs may wrongly mistaken it as a DSD data, and since all '0' mean maximum negative output voltage thus generating a loud pop.

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Some people preferred the basic level Spring in NOS mode compared to other reference DACs and the inverse in OS mode, and most agree the Spring sounds best in NOS mode. Do you know if the Spring has an analog low-pass (reconstruction filter), and what type of filter it is (curious if it is a sharp filter between 20 and 22.05 with ripples or something more gentle that would work well if upsampling in the computer)?

I don't care much about DSD, and would probably not use OS. Many may think the same. In that case wouldn't it be better to transfer the cost of something that a lot of people may not use to a better USB implementation? Or remove the USB completely and sell it as a separate module...

 

I did post this question very early on in the thread but there was no response to it. So I contacted Tim from Kitsune regarding on NOS at 44.1k, he got reply from Jeff Zhu that there will 44.1k sampling noise coming out, the LPF in analog stage can reduce it but not very much but he assures me that it will not cause problems (instability) in amplifier and speakers.

 

The purpose of over-sampling is to shift 44.1kHz all the way to 352.8kHz (assuming it uses 8x over sampling). At 352.8kHz, it can be easily filter with a less steep digital filter. The first image and subsequent images can be greatly reduced as a result of over sampling and a gentle digital filter. This also improves the SNR and it will definitely look good on technical performance since now 44.1k is shifted much further away from the audio band, i.e, 20kHz. However, if you started with a Hi-Res like 192kHz or even 352.8kHz is lesser issue since the sampling frequencies are high enough to begin with.

 

The downside of over-sampling and digital filter is it creates an artifact called 'ringing' which can severely affect the impulse response. Since a good impulse response is responsible for accurately reproduce many of musical instruments; its 'timbre' and 'natural tone' There's an article on this, you can read this:

6moons audioreviews: Metrum Acoustics Pavane

 

Hope this helps.

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Thanks @ted_b, what I like is I can simply bypass the built-in over-sampling and digital filter and use a software base like HQP or the latest Roon 1.3 with up-sampling DSP to tailor the sound I like.

 

Most DACs in the market has little provision to allow the user to bypass the hardware over sampling and digital filter inside the DACs (especially those use off selves DAC chips) so even having a good software base up sampling like HQP, one is already severely limited by the built-in hardware over sampling and digital filter. This is main reason I brought Holo Spring DAC and I simply love the sound of this DAC!

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Thanks @ted_b, what I like is I can simply bypass the built-in over-sampling and digital filter and use a software base like HQP or the latest Roon 1.3 with up-sampling DSP to tailor the sound I like.

 

Most DACs in the market has little provision to allow the user to bypass the hardware over sampling and digital filter inside the DACs (especially those use off selves DAC chips) so even having a good software base up sampling like HQP, one is already severely limited by the built-in hardware over sampling and digital filter. This is main reason I brought Holo Spring DAC and I simply love the sound of this DAC!

 

Bingo!

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Thanks for the reply. I was asking about the analogue filter. If it was designed as a NOS DAC normally it would be a filterless DAC or have a relatively sharp filter. From your reply it seems to have an analogue filter with a gentle slope (probably designed for upsampling in the DAC or in the computer), that works well with high res files or upsampling in the computer. This brings another issue in my case, 176.4, 192, 352.8, 384 would have to go through USB, and if the USB implementation in the DAC isn't good enough that is another can of worms.

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Axle_69, if you are so worried about the USB input (not sure why, myself and others love it, given enough care with the input stream) then all other digital inputs support 192k and DSD64 (DoP), and the I2S supports everything (but DSD512 in my setup).

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