HOLO Audio Spring DAC - R2R DSD512

[guymrob]

The AKM SRC inside DAC can also upsample to max DSD256 by selecting OS DSD mode. Has anyone compare this with HQPlayer?

[guymrob]

I don't think hardware manufacturers especially small start ups like Holo Audio Springs are willing to incorporate MQA because of expensive licensing deals. Besides, the over-sampling mode will simply destroy the MQA code render it undecoded in the end. Don't waste your time talking about MQA here. The whole ideas about this particular product here is this unique R2R can do NOS/OS PCM/DSD and ability to support high sampling rates both in PCM and DSD.

[guymrob]

Thanks Ted_B, is the latest unit(firmware) were able to DSD512 natively using latest Windows driver provided by Holo Audio Springs? I heard in the forum about PCM 768k but not sure this is supported.

[guymrob]

The NOS mode is the most talk about in this forum. When playing back a low sample like 44.1k without OS, does this present itself more noise above the 22k spectrum? Most off selves DACs from manufacturers used a default 8x OS to shift the sampling noise to higher band thus making it easier to filter off the noise.

[guymrob]

This is an unresolved issue especially using an USB-DAC; the slight click when playing the next DSD track is obvious. The only way to avoid that is to use I2S interface or use a built-in DAC together with streamer, the interface between the DSP and DAC chip is via I2S allows precise control over signals mute without delay or latency.

[guymrob]

I've two experiments, both using an Auralic Aries Mini streamer; one using the internal DAC, a ESS9018KM and the other is routed via USB to an external USB- DAC. When I switch using the internal DAC, playing back DSD, i can't hear any clicks at all and I noticed music kind of fade out when start playing the next track or switch tracks. However, when I switch to external USB-DAC, I can hear clicks when start playing next track or switch tracks. Obviously the built-in DAC chip have much control over mute delay. The Internal CPU can be programmed to send mute signal to gate the analog outputs either in the form solid-state switch or a relay.

[guymrob]

I just ordered the Holo Spring Audio DAC, does anyone know where can I download USB driver for Windows 10?

[guymrob]

Cool, thanks!

[guymrob]

I came across new firmware update for Holo Audio Spring DAC... Please proceed with caution as incorrect flashing of firmware will render the USB input not working!!!

The latest firmware is 16.92, you need to install XMOS-Stereo-USB-Audio-Class2-Driver-3036(v2.23.0).exe in order to check which firmware is currently having (I will check it later) There's a document: 泉USB固件升级方法.pdf which show how to check and flash the USB XMOS firmware. Can anyone check with Jeff Zhu, the designer of Holo Audio Spring DAC regarding on this before proceeding and what is this firmware suppose to fix?

 

https://www.shenzhenaudio.com/holo-audio-spring-r2r-dac-hi-end-dsd-dac-220v-only.html - USB firmware update

spring_1692_update.bin - 哔哩网盘搜索 - 速度最快资源最多的百度网盘搜索神器 - another website on USB firmware update.

[guymrob]

Wow, why the alert? This firmware (1692) has been since last September. It is the most recent but surprised anyone would not already have it installed from Tim/factory.

 

Thanks Ted, I've check mine (I got it couples of days ago) and it is indeed 1692. Looks like updating the USB input firmware can be done without sending the unit to the factory!

[guymrob]

Hi Ted,

 

I noticed when I setup Auralic Aries (presumed it uses some kind of Linux based OS), I got click noise when switching from DSD64/DSD128 back to PCM or vice-versa. The noise becomes quite apparent like pop noise when switch from PCM or DSD64/DSD128 to DSD256 and back to it again. Every time when I try to play DSD256, I've mute the volume. Is there a way to fix this problem?

 

To my surprised, this issue is non-existent if I use Roon+Windows with the supplied Holo Spring certified Windows 10 drivers.

[guymrob]

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.

[guymrob]

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.

[guymrob]

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.

[guymrob]

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.

[guymrob]

I've tried using Windows+Roon with Holo spring Windows driver, I don't experience any loud pop at all, just clicks with DSD but with Auralic Aries Mini, there's a loud pop sound!

[guymrob]

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.

[guymrob]

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!

[guymrob]

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.

 

@axel_69, may I know what issue you encountered using the built-in USB input? The built-in USB (with XMOS U8) board is made by Holo Spring (Jeff Zhu), you can see the name and the Holo Spring logo printed on the USB PCB board. This is NOT an off selves or OEM USB input board where you can just find anywhere else.

 

Jeff Zhu spent a lot of his effort optimising the USB driver (Thesycon and Windows signed driver!) plus latest firmware so that it can work reliably on Linux and Windows.

[guymrob]

I didn't say I had. Trying to make a decision between the Spring and another DAC. It seems some people feel the need to put something like the Singxer SU-1 DDC in the middle. In the case of the other DAC it would be computer without upsampling > Toslink up to 96 > upsampling in the other DAC, whereas in the case of the Spring it would have to be upsampling in the computer > USB > ? > Spring. The Spring may take the lead on the DAC itself, the other DAC will take the lead on the upsampling algorithm (probably even compared with upsampling in the computer), but what happens in the middle is important as well. I could upsample 44.1 to 88.2 and use Toslink with the Spring but my guess is that the analogue low pass has a gentle slope and a higher sampling rate will be better. So it ends up being a question about upsampling algorithms and digital connections as well. How does the USB input (with output directly from the computer) compares with the others?

 

Thanks for your reply, first let me put things in a simpler way. There are people who use Singxer SU-1 reported some improvement in sound quality, for me I did hear a minor different; I feel the background is a little 'darker' but that doesn't mean the built-in USB is inferior. In fact, the difference is so small it doesn't warrant me to upgrade to Singxer SU-1 at all!

 

Now, I don't just listen to USB input but I also have a CD player that hooked up to the coaxial digital input too. When I compared 44.1k from the CD and one from a PC, I can't tell the difference between both of them!

 

Regarding on the different type of over sampling and digital filter, whether it is hardware or software, the hardware over sampling and digital filter will never sound the same from different manufacturers; because either they use a different type of chip or the over sampling and digital filter have a different algorithm. When you buy them, you have to like the signature of the sound!

 

For software up sampling and digital filters, all these can be customised and tailored to your liking, the flexibility of software is simply out of this world.

 

In order to do that, the DAC must have a provision to bypass the hardware over sampling and digital filter, so software up sampling and digital filter will really shine. The Holo Spring give user a lot of flexibility not found in most high end DACs. By doing that, people who are 'purists' who just want to listen directly without any form of over sampling can do so (better impulse response) or can use the hardware or software up sampling to suit their likings.

[guymrob]

This is incorrect. For instance, "most DACs" use an 8x first stage filter in their oversampling engine. If the DAC in question can receive a 352.8 sample rate (which is 8x red book, 44.1) and if you oversample in software to 352.8 and send that to the DAC, you will be bypassing the first stage digital filter. Additionally, most audible "problems" of a poor oversampling algorithm occur during the first pass of the oversampling process (any artifacts of the next stage(s) of oversampling will be entirely out of the audible bandwidth). You do not need a DAC which can be set to a NOS mode to enjoy the benefits of software based oversampling.

 

Most of the off selves DACs have 8x oversampling filters that are consisted of three 2x filters.

 

When you input data at 44.1/48 (single rate), you get all the three filters. When you input data at 88.2/96 (double rate), you get two filters. And when you input data at 176.4/192 you get only one. So, in all cases, final rate is capped at 352.8/384. So you are right to say if you already up sample in the software to 352.8/384, the DAC will auto select to no over sampling since it is already capped in the final stage. However, there are custom design DAC that uses FPGA have the processing power to over sample at higher rate, like 705/768 or even higher in the MHz range even when input sample is 352.8/384, therefore there is still over sampling going on. To add it further, some even convert PCM to DSD internally! It all depends on the processing power given by chip.

 

By using a NOS DAC, you can be 100% sure that there is no hardware over sampling and digital filter come into play, just a gentle LPF in the analog stage. Over sampling and digital filter is then done on the software level, like HQP, giving users more flexibility and better quality.

[guymrob]

As I mentioned, yes, often there is still more OS beyond 352.8/384. But, going from 352.8/384 to (whatever, often MHz rage) will generally not be audible, as any artifacts produced by this OS step are so far beyond the audible range as to not matter, where the first OS step, from (for example) 44.1 to 352.8 is very audible. So, my point stands that one not need an actual NOS DAC (or setting in their DAC) to experience the difference of software based OS.

 

If 44.1 to 352.8 is very audible in hardware over sampling inside in a DAC then obviously you want to remove this and apply external software based OS. After all, you don't need to have both hardware and software OS. And since people use HQP to do OS because it sound much better than those built-in hardware OS, obviously bypass it will make software OS really shine. Besides, bypassing it actually shortened the signal path...

[guymrob]

It depends on your analog filters. It can still have audible impact due to correlated intermodulation products as explained many times in other threads. Only if you have good and steep enough analog reconstruction filters that remove all the images at 352.8/384k and multiples, you don't have such worries. But so far most of the DAC's I've measured still leave quite a bit of images...

 

If you start from 44.1/24 with 8x oversampling, you need analog filter that rolls off to -144 dB by 330.75 kHz. IOW, you need about 36 dB/octave (6th order) analog filter. And then you need to forget hires, because it needs to begin to roll-off already at about 25 kHz. As a result, the phase response will look pretty bad too.

 

Correct me if I'm wrong, my understanding is hires recording at 352.8/384k (DXD) begins to roll of before half the Nyquist frequency, in this case 176.4/192k. I believe modern digital filter will able to take care the roll off beginning at 176.4/192K all the way down to 352.8/384k, but I'm not sure what is the attenuation factors these digital filters has, how many dB/octave? I'm sure nobody wants want to build a analogue filter to do that, as you mentioned above.

 

If over sampling digital filter is going to solve the 352.8/384k (DXD) while maintaining the hires content, then the final output stream is going to be 2.822/3072MHz, 8x OS of 352.8/384k. Not all modern DACs are designed to accept such a high data rate, especially the off selves DACs due to power consumption and heat issues.

[guymrob]

Can you tell me what is different between digital filter in the over sampling block vs the analogue filter right after being converted to analogue? I have an impression that over sampling digital filter has a much greater than -3dB roll off/octave, depending whether is a steep or gentle roll off. Thanks for valuable inputs.

[guymrob]

@Bimmer100, do you have any good news for us on the native DSD loud pop problem? I think you mentioned somewhere Jeff Zhu was looking at implementing a fix/workaround in the DAC. Any update? Can end users apply such an update or will it require sending the DAC to you or Jeff?

 

It's been said this problem doesn't happen with the Windows driver. Is it because the driver contains a workaround? Can it be added to the Linux driver to provide relief for the microRendu users?

 

I've contacted Jeff Zhu on my Auralic Aries Mini(Linux based OS) containing a loud pop when Aries Mini transport DSD256 natively (default), while DSD64/DSD128 is transported via DoP mode by Aries (default) so I don't experience the loud pop. He is looking at it. I also contacted Auralic, they too is investigating on the issue. It can be Holo Spring or the Auralic but I've to wait to get further reply from them.