HOLO Audio Spring DAC - R2R DSD512

[Superdad]

I see that Holo uses Intel/Altera FPGAs—on both their main board and on the USB input board.  There is a severe global shortage of Altera MAX FPGAs as Intel is using their production spots (with their fab house partners) for larger processors and CPUs. Lead times for Altera. FPGAs now stretch into 2024.

Unless Holo bought enough stock to carry them forward, they may soon need to pause, learn a different development platform, and rewrite all their code to suit a more readily available FPGA series from a different manufacturer. And that takes time.
 

This is what happened to UpTone recently—which forced the discontinuation (for now) of our popular UltraCap LPS-1.2 and has slowed down design work on a important new USB product we have been developing for nearly a year. [We are in the process of shifting to the small, innovative FPGA-focused firm Efinix (https://www.efinixinc.com/)?]
 

The Holo Spring 3 looks terrific, but those on the fence may be wise to get one sooner than later… 

[DuckToller]

2 minutes ago, GoldenOne said:

I am!
For the money it's a crazy good option IMO.

L1/L2 spring 3 have the 'normal' usb card. Still has galvanic isolation, but no titanis chip and not as advanced. How much of a difference that makes I cannot say as I've not compared the two.

I said $2500 cause L2 is $2500. (L1 being cheaper and KTE being a bit more).
Having tried the KTE and L2 may though, there is a very subtle difference but not enough to change anything I said in my may video for example.

The spring 3 l1/2/3 have some slight component upgrades but they're the same design, same dac. It's not going to make it a drastically better/worse product or anything.
 

 

From an USB point of view, because you desribed it as "the way to go" feeding the Spring 3 - without a DDC - did that refer explicitely to the model you used?
Let's put it this way: If there's no need for a distinct DDC in front of that USB entry called Titanis 2.0 of the KTE version (according to your impressive Jitter measurement), the difference in budget may be well spent (thinking as well about the abundance of non-cheap USB cables ... Chord alike ;-)
However, even my ears haven't identified Jitter as a probleme for a long time ... maybe Titanis 2.0 could be the ultimate way to go?
You did not hear/measure/compare the S3-L2 to S3/L3 USB inputs against each other, did you?
Would be great to see if any advancement shown there,  are essentially different from no-titanis USB solution.
I guess, Kitsune could love that kind of discussion ... some of us audiphiles will have that nagging feeling that they missed an opportunity with L2 ... personally I think it is a question of the use case and I am happy to have many choices in regard of a product like the Spring 3
Thus, I think it is valuable to define precisely the pros and cons of a serious plus in investment.
btw the cable thing is only intellectual pastime, I am not at all an advocate of expensive cabling ...

[Ozan Bolat]

@Miska :

are the following settings correct to try NOS ? including letting Dac bits to default instead of 20 when I tried 1536 ? 

will convolution still be applied and if yes correctly so (remember reading should be paired with upsampling by a value of x2 at least)?

When oversampling  my choice is SDM. I see @GoldenOne uses sinc Mx I have never been fond of, I'm used to use Sinc S and ext2 based on content since our last discussion on the topic but reassessed gauss long I will further investigate ; what is your current opinion ? BTW @GoldenOne does not discuss PCM vs SDM oversampling in HQP for the S3 and I see he offers settings limited to 128 where it should be 256, cpu allowing

Guess you're not recommending NOS but where would you put the technical threshold below which roll off or other phenomenon impair reproduction ? Are we safe from such impairments above 88.2?

[dankef]

Thanks @GoldenOne for the excellent review. A few questions:

 

1. You spoke highly of the preamp option for the price. Did you have a view about how it compares with the Serene or perhaps the preamp in the Azure? I like the idea of one box although it would be ideal if there were analog inputs to use with a turntable down the line (similar to the Rockna Wavelight).
2. Do you have a view about how the USB (for me, out of a MacBook Air or Mac Mini (M1) via dongle or hub) compares with i2s or AES out of a source like the Pi2AES?
3. Did you compare the SE outs with the XLR outs?

My intention would be to run the Spring 3 single-ended into a tube amp and also via XLR into active monitors. 
 

Many thanks.

[Miska]

4 hours ago, Ozan Bolat said:

@Miska :

are the following settings correct to try NOS ? including letting Dac bits to default instead of 20 when I tried 1536 ? 

will convolution still be applied and if yes correctly so (remember reading should be paired with upsampling by a value of x2 at least)?

 

No, (assuming you also change to PCM output) if you perform any DSP, like volume control, convolution, or anything else, you must set some proper dither. Otherwise you have a distortion generator. And well, doing NOS that way you have a distortion generator anyway since imaging distortion will be in tens of % range. But without proper dither you have two distortion types instead of just one. (not counting IMD and TIM resulting from excessive imaging since these depend more on your amplifier gear)

 

Quote

When oversampling  my choice is SDM. I see @GoldenOne uses sinc Mx I have never been fond of, I'm used to use Sinc S and ext2 based on content since our last discussion on the topic but reassessed gauss long I will further investigate ; what is your current opinion ? BTW @GoldenOne does not discuss PCM vs SDM oversampling in HQP for the S3 and I see he offers settings limited to 128 where it should be 256, cpu allowing

 

Filter choice is up to your personal preferences. I tend to use shorter filters, like poly-sinc-ext2, poly-sinc-short-mp(-2s) and poly-sinc-gauss/poly-sinc-gauss-long.

 

Quote

Guess you're not recommending NOS but where would you put the technical threshold below which roll off or other phenomenon impair reproduction ? Are we safe from such impairments above 88.2?

 

No, you begin to be safe around 1.5M PCM rates.

 

[Ozan Bolat]

13 minutes ago, Miska said:

 

No, (assuming you also change to PCM output) if you perform any DSP, like volume control, convolution, or anything else, you must set some proper dither. Otherwise you have a distortion generator. And well, doing NOS that way you have a distortion generator anyway since imaging distortion will be in tens of % range. But without proper dither you have two distortion types instead of just one. (not counting IMD and TIM resulting from excessive imaging since these depend more on your amplifier gear)

 

 

Filter choice is up to your personal preferences. I tend to use shorter filters, like poly-sinc-ext2, poly-sinc-short-mp(-2s) and poly-sinc-gauss/poly-sinc-gauss-long.

 

 

No, you begin to be safe around 1.5M PCM rates.

 

Hope your post gets framed thanks to pluri thanks and likes ; important one.

(I did not change output for I did not even bother to try before your answer)

[Miska]

On 9/22/2021 at 12:40 PM, 87mpi said:

I find the PCM part better, more alive and real, of the DSD part. I feed the dac with PCM 1536 Sync-M and LSN15. I used to use DSD 256 7EC Native.

 

To me, to PCM side sounds flat on transients (a bit "flap-flap"). It lacks the body, weight, depth and kick of DSD side running at DSD256 with ASDM7EC (88.2k Daft Punk). Also on dense mixes the PCM side can become congested (24-bit Opeth - Pale Communion, 96k Steven Wilson/Porcupine Tree/Storm Corrosion). This is common to any R2R I've heard so far. But at 1.5M PCM this effect is the least.

 

Just remember to adjust for the 6 dB volume difference between the two when comparing.

 

P.S.Now it is getting closer to time of my favorite fall time song - Opeth's Eternal Rains Will Come, from their Pale Communion album.

 

[Bertel]

3 hours ago, Miska said:

 

Opeth's Eternal Rains Will Come, from their Pale Communion album.

 

❤️

[Cogito]

@GoldenOne

 

Thanks for the excellent review of Spring 3.

 

Couple of small issues:

 

1. DSP volume attenuation not only reduces the headroom, but also reduces the resolution of the source material. 

Redbook 16 bit PCM has the headroom of 16 x 6 = 96 db

The max value that can be represented by a 16bit binary number is 65,536. 

DAC's reference voltage (5V?) is divided into 65,536 equal parts to represents all possible values of 16bit binary (PCM sample). Each step is 0.015mv.

 

If the volume is attenuated by 6dB in the digital domain reduces the bit depth to 15.

Redbook 15 bit PCM has the headroom of 16 x 6 = 90 db

The max value that can be represented by a 15bit binary number is 32768. 

DAC's reference voltage (5V?) is divided into 32767 equal parts to represents all possible values of 16bit binary (PCM sample). Each step is 0.03mv. This is much coarser representation that the original redbook source. That result is, you would loose microdynamics and resolution. For this reason audiophiles never use DSP for volume attenuation.

 

2. Preamp compensates for the headroom loss due to DSP attenuation.

This is not true. Preamp has two purposes:

1. Volume attenuation in the analog domain with potentionmeters

2. Analog signal amplification

 

Preamps and Power amps do not change the headroom present in the source material. Preamp can increase the volume at with the audio is played but it does not change the headroom.  If the DSP attenuation reduces the resolution from 16bit to 14bit, preamp can compensate for the loss of volume but the headroom will still be that of of 14bit.

 

 

 

 

 

 

[GoldenOne]

10 minutes ago, Cogito said:

@GoldenOne

 

Thanks for the excellent review of Spring 3.

 

Couple of small issues:

 

1. DSP volume attenuation not only reduces the headroom, but also reduces the resolution of the source material. 

Redbook 16 bit PCM has the headroom of 16 x 6 = 96 db

The max value that can be represented by a 16bit binary number is 65,536. 

DAC's reference voltage (5V?) is divided into 65,536 equal parts to represents all possible values of 16bit binary (PCM sample). Each step is 0.015mv.

 

If the volume is attenuated by 6dB in the digital domain reduces the bit depth to 15.

Redbook 15 bit PCM has the headroom of 16 x 6 = 90 db

The max value that can be represented by a 15bit binary number is 32768. 

DAC's reference voltage (5V?) is divided into 32767 equal parts to represents all possible values of 16bit binary (PCM sample). Each step is 0.03mv. This is much coarser representation that the original redbook source. That result is, you would loose microdynamics and resolution. For this reason audiophiles never use DSP for volume attenuation.

 

2. Preamp compensates for the headroom loss due to DSP attenuation.

This is not true. Preamp has two purposes:

1. Volume attenuation in the analog domain with potentionmeters

2. Analog signal amplification

 

Preamps and Power amps do not change the headroom present in the source material. Preamp can increase the volume at with the audio is played but it does not change the headroom.  If the DSP attenuation reduces the resolution from 16bit to 14bit, preamp can compensate for the loss of volume but the headroom will still be that of of 14bit.

 

 

 

 

 

 

The issue is that whilst most music might be 16 bit, the DAC in most modern examples will have THD, SNR and AES17 Dynamic range higher than that, and oversampling/dsp volume control will likely be being done at a higher bit depth than the source material too. Which is why 96dB isn't enough to call something 'perfect'.

Dithering and noise shaping also have enormous effects

Additionally, I understand that the preamp doesn't affect the source material, but it is part of the playback chain. If you have a DAC with say 120dB dynamic range, and you attenuate 10dB in DSP, you will lose 10dB dynamic range (in most cases, some dacs with non-linear THD+N/level performance might be slightly different).

But if you have say a preamp with 140dB dynamic range. Now, you could attenuate by 10dB, and it's entirely likely that the noise floor of the preamp is still well below that of the DAC. Meaning you might still have the full 120dB dynamic range as you'd be limited by the DAC not the preamp.

I did a few tests with this on the spring 3 and found that I got about 5dB extra headroom before dynamic range was affected (and the spring 3 has exceptionally high dynamic range anyway so could attenuate further before dynamic range reaches the level of many other similarly priced dacs).

The Holo Serene is probably the most drastic example of this due to the quite frankly ridiculous dynamic range it has.
I'll take it out of my chain and do some practical demonstrations of the above in the next few days. Need to wait for a new headphone amp to arrive then I can take the serene out and put it on the measurement bench

[Cogito]

8 minutes ago, GoldenOne said:

The issue is that whilst most music might be 16 bit, the DAC in most modern examples will have THD, SNR and AES17 Dynamic range higher than that, and oversampling/dsp volume control will likely be being done at a higher bit depth than the source material too. Which is why 96dB isn't enough to call something 'perfect'.

 

Different issues. The specs like THD, SNR provide by DAC measure only the artifacts introduced by the DAC. DACs are not enhancing the digital source.

 

13 minutes ago, GoldenOne said:

Additionally, I understand that the preamp doesn't affect the source material, but it is part of the playback chain. If you have a DAC with say 120dB dynamic range, and you attenuate 10dB in DSP, you will lose 10dB dynamic range (in most cases, some dacs with non-linear THD+N/level performance might be slightly different).

But if you have say a preamp with 140dB dynamic range. Now, you could attenuate by 10dB, and it's entirely likely that the noise floor of the preamp is still well below that of the DAC. Meaning you might still have the full 120dB dynamic range as you'd be limited by the DAC not the preamp.

 

You are explaining  passive attenuators, which audiophiles stay away from as they compromise micro dynamics.

In a typical active preamp, about 90% of the original signal is burned off and amplified again. In our original example, 14bit PCM has a dynamic range has a noise floor of -84dB.  In the preamp and amp, all analog signals, which includes noise, is also amplified.  Preamp having lower noise floor only means preamp is not adding noise to the signal.

 

18 minutes ago, GoldenOne said:

I did a few tests with this on the spring 3 and found that I got about 5dB extra headroom before dynamic range was affected (and the spring 3 has exceptionally high dynamic range anyway so could attenuate further before dynamic range reaches the level of many other similarly priced dacs).

 

I have no issue with the DAC itself. I am pointing out the evils of DSP volume attenuation.

 

[Miska]

10 minutes ago, Cogito said:

1. DSP volume attenuation not only reduces the headroom, but also reduces the resolution of the source material. 

Redbook 16 bit PCM has the headroom of 16 x 6 = 96 db

The max value that can be represented by a 16bit binary number is 65,536. 

DAC's reference voltage (5V?) is divided into 65,536 equal parts to represents all possible values of 16bit binary (PCM sample). Each step is 0.015mv.

 

Let's say you output at DSD from HQPlayer. This means you have well over 160 dB dynamic range in RedBook frequency range. This leaves you volume adjustment range of at least 64 dB without any loss in resolution. It also means that the digital noise floor is always some 40+ dB below your achievable analog noise floor.

 

Let's say you use my recommended settings of 20-bit output and suitable noise-shaper with PCM output instead. This gives also about 160 dB dynamic range in RedBook frequency range.

 

The internal DSP pipeline in HQPlayer has SNR of over 300 dB and dynamic range in thousands of dB.

 

10 minutes ago, Cogito said:

1. Volume attenuation in the analog domain with potentionmeters

2. Analog signal amplification

 

Preamps and Power amps do not change the headroom present in the source material. Preamp can increase the volume at with the audio is played but it does not change the headroom.  If the DSP attenuation reduces the resolution from 16bit to 14bit, preamp can compensate for the loss of volume but the headroom will still be that of of 14bit.

 

Preamp should be called pre-att these days. It will always add noise and distortion. Analog domain is always very restricted when compared to what can be done in digital domain where you don't have limitations like thermal (Johnson-Nyquist) noise.

[GoldenOne]

5 minutes ago, Cogito said:

Different issues. The specs like THD, SNR provide by DAC measure only the artifacts introduced by the DAC. DACs are not enhancing the digital source.

You can convert 16 bit source info to 24 bit to apply DSP with less loss of accuracy/dynamic range.
There is no reason you have to keep 16 bit as 16 bit (and internally the vast majority of dacs will not)

 

 

6 minutes ago, Cogito said:

You are explaining  passive attenuators, which audiophiles stay away from as they compromise micro dynamics.

In a typical active preamp, about 90% of the original signal is burned off and amplified again. In our original example, 14bit PCM has a dynamic range has a noise floor of -84dB.  In the preamp and amp, all analog signals, which includes noise, is also amplified.  Preamp having lower noise floor only means preamp is not adding noise to the signal.

No I'm not. The serene is an active preamp.
https://www.l7audiolab.com/f/measurements-of-holoaudio-serene-preamplifierpre-retail/

 

As mentioned, I'll do a practical demonstration showing the effect of dynamic range in a preamp in the next few days.

 

[Cogito]

@Miska

 

i agree with you about HQP< but that’s beside my point.

[Cogito]

5 minutes ago, GoldenOne said:

You can convert 16 bit source info to 24 bit to apply DSP with less loss of accuracy/dynamic range.
There is no reason you have to keep 16 bit as 16 bit (and internally the vast majority of dacs will not)
 

I am addressing the statements you made in the review. If you mentioned the above in you review, this discussion would not exist.

[Luis Cesar Saiz]

Hello there !!.  I'm mostly a vinyl player, but I want to return to my CD collection and listen to Qobuz too (streaming from PC or Phone) .
 

I'm going to purchase the Spring 3 level 3 and I have a ton of CDs that I want to play them with my current CD player (A Panasonic UB9000 4K/BluRay player). Which method is "best" to use it as a transport.
- Coax output to coax input on SP3 
- HDMI (audio out) to i2s on SP3 (since the Spring can accommodate different pin-out ... but not sure if it works as is as simple as I put it here... :D )
- Buy a Coax to USB converter or a streamer that accept coax input...

My CD collection is ripped onto a USB hard disk connected to the Pana, so I will do some comparisons... :D

On a proper and modern CD Transport, any of you has experience on Jay's Audio CDT2 or Project's CD Box RS2 T or PS Audio,  both i2s, I haven't seen any transport with USB out... or any other transport recommendation. (I have no longer my Mark Levinson 39 so... )

Best
Luis

[Miska]

3 hours ago, Luis Cesar Saiz said:

- HDMI (audio out) to i2s on SP3 (since the Spring can accommodate different pin-out ... but not sure if it works as is as simple as I put it here... :D )

 

Note that the I2S input Spring 3 is not HDMI. It uses same connector and cable, but that's the only common thing. Do not connect HDMI devices to it!

 

[Cogito]

4 hours ago, Luis Cesar Saiz said:

Hello there !!.  I'm mostly a vinyl player, but I want to return to my CD collection and listen to Qobuz too (streaming from PC or Phone) .

if you want your CD collection to sound as good as vinyl this is what you need to do in addition to Spring 3.

 

Step 1:

Eliminate/minimize noise in the signal getting to DAC.

1. Use only wired network connection for the music player PC.
2. Use quality network card (like JCAT) or Fiberoptic network card with Ethernet to optical converter 
3. Use quality USB card (like JCAT)

 

Step 2:

To fully realize the potential of Spring, use a quality up sampler like HQPlayer in the music player

 

Step 3:

Digitize you SACD and CD collection and stream them with roon/Audirvāna (hosted on another pc). 

 

 

[Ozan Bolat]

@Miska With your Spring that are level 2 do you use a NUC or any kind of NAA configuration or the USB optical decoupling inside the Holo is good enough so it's not useful to have an intermediate light working computer? I think I have to keep my Up anyway to be able to do 1536 that my Mac would not allow, correct ? BTW is there any technical reason linked to recording or mastering process that would justify that sometimes PCM 1536 beats SDM 256 ? Or is it only my mood sometimes? This morning for instance the sense of Live (like a fisheye view of the clamouring crowd at the beginning of Fragile) was much much better with SDM and Sting ; and with MFSL U2's With or without you drumming and bass notes were much more exciting and simply real with SDM and PCM simply failed. Remember you recommend lphase for space and mp for transients and in the 2 examples above, SDM beats PCM on both space and transients while PCM might shine with grounding of a position in the soundstage and making the sound heavier, aspects I sometimes prefer depending on source and mood

[Miska]

1 hour ago, Ozan Bolat said:

@Miska With your Spring that are level 2 do you use a NUC or any kind of NAA configuration or the USB optical decoupling inside the Holo is good enough so it's not useful to have an intermediate light working computer?

 

I think there are still some benefits from a NAA, but certainly isolation in Spring helps.

 

My Spring 1 is now in storage, not in use at the moment.

 

Spring 2 is directly connected to my Linux Xeon workstation and working as a DAC on my desktop headphone system.

 

Spring 3 is at the office equipment rack, used through Up-based NAA powered by Ferrum Hypsos PSU.

 

Quote

I think I have to keep my Up anyway to be able to do 1536 that my Mac would not allow, correct ?

 

Exactly. Although you can get 1.5M from Mac if you set DAC Bits to 16. At least this worked on Spring 2. If the Spring 3 also supports 16-bit transfer mode then it should work with it too.

 

Quote

BTW is there any technical reason linked to recording or mastering process that would justify that sometimes PCM 1536 beats SDM 256 ?

 

No, not really.

 

But if you want pure PCM recordings that you can try on a pure PCM DAC, you can look for 176.4/24 recordings by Reference Recordings, made with Pacific Microsonics Model 2 ADC. At least the older ones, not sure if their converter is still alive and operational. These don't need an apodizing filter either. Works well for SDM outputs too! Their new material seems to be 192/24 or 96/24 which is from some SDM ADC.

 

I think Mobile Fidelity also has had some similar recordings, but I'm not sure (key is that it at least needs to 176.4/24 otherwise it is certainly not).

 

For pure DSD recordings you can try on a pure DSD DAC, you can take a look at NativeDSD.

 

Since the content and recordings are different, not suitable for comparing formats though.

 

Quote

Or is it only my mood sometimes?

 

Subjectively one may of course like either way. I can understand for example that in some cases, PCM with more collapsed image / sound stage could sound more intimate and preferred. It could depend on the particular recording too.

 

[Ozan Bolat]

5 minutes ago, Miska said:

 

I think there are still some benefits from a NAA, but certainly isolation in Spring helps.

 

My Spring 1 is now in storage, not in use at the moment.

 

Spring 2 is directly connected to my Linux Xeon workstation and working as a DAC on my desktop headphone system.

 

Spring 3 is at the office equipment rack, used through Up-based NAA powered by Ferrum Hypsos PSU.

 

 

Exactly. Although you can get 1.5M from Mac if you set DAC Bits to 16. At least this worked on Spring 2. If the Spring 3 also supports 16-bit transfer mode then it should work with it too.

 

 

No, not really.

 

But if you want pure PCM recordings that you can try on a pure PCM DAC, you can look for 176.4/24 recordings by Reference Recordings, made with Pacific Microsonics Model 2 ADC. At least the older ones, not sure if their converter is still alive and operational. These don't need an apodizing filter either. Works well for SDM outputs too! Their new material seems to be 192/24 or 96/24 which is from some SDM ADC.

 

I think Mobile Fidelity also has had some similar recordings, but I'm not sure (key is that it at least needs to 176.4/24 otherwise it is certainly not).

 

For pure DSD recordings you can try on a pure DSD DAC, you can take a look at NativeDSD.

 

Since the content and recordings are different, not suitable for comparing formats though.

 

 

Subjectively one may of course like either way. I can understand for example that in some cases, PCM with more collapsed image / sound stage could sound more intimate and preferred. It could depend on the particular recording too.

 

Thank you very much, very thorough and informative. But now I'm troubled... Ferrum Hypsos PSU cost 1195 euros ! is that what is required to make a Up NUC worthy !?

this link adds sources for Pacific Microsonics Model 2 ADC made recordings or transfers

 

 

[Miska]

11 minutes ago, Ozan Bolat said:

Thank you very much, very thorough and informative. But now I'm troubled... Ferrum Hypsos PSU cost 1195 euros ! is that what is required to make a Up NUC worthy !?

 

I used it with 20€ Meanwell PSU before, but changed it to Hypsos recently.

 

You can use wide variety of different PSUs at different price points with it. It doesn't mean that you would need to use something like Hypsos for it.

 

[Ozan Bolat]

3 hours ago, Miska said:

 

But if you want pure PCM recordings that you can try on a pure PCM DAC, you can look for 176.4/24 recordings by Reference Recordings, made with Pacific Microsonics Model 2 ADC.

 

Can not provide a guaranteed list of albums but the 100 or so Pacific Microsonics Model 2 ADC produced have been in heavy use and still are in some studios : I will stick to the practice of giving both PCM and SDM a try for each album I want to seriously listen to. The list of pure PCM recordings is probably quite large indeed but there's also some logic to find that the SDM path better suits recent recordings or transfers

[Ozan Bolat]

On 10/5/2021 at 2:45 PM, Miska said:

 

 

For pure DSD recordings you can try on a pure DSD DAC, you can take a look at NativeDSD.

 

 

 

Music first, SQ second : I have never been interested in Jazz at the Pawnshop kind of stuff... It's not pure for it started being analog recorded but Bill Evans 2xHD DSD 128 offers sound incredibly good

[bodiebill]

Last week I compared the Holo Spring 3 Level 2 with the Holo May KTE at Magna Hifi, see:

 

 

Against general preference I preferred the Spring 3 L2 to the May KTE. It could have been the specific room and setup, or it could be just me. The dealer did admit that the room sounds overly bright and that the May KTE tends to emphasize that.

 

Now my question:

 

Suppose I would order a Spring, I find it a difficult to decide for or against the preamp module. Actually I find preamps one of the most confusing things in audio: passives, actives, digital, volume vs gain, attenuation, etc. Also, most preamps I heard (not that many I must admit) for me seriously deteriorated the sound. Some time ago I ended up with a The Truth (Hornshoppe) passive preamp and I loved it. May I use the word 'transparant'? A device you can just set and forget.

 

However, when I went the Dirac way and purchased an (all-digital) MiniDSP Studio, I briefly wanted to try its digital 'preamp', expecting to hear the deterioration I always heard when using digital vc's. But no! It surprisingly sounded great and transparant to my ears, and now I am using the MiniDSP as my preamp as attenuator instead of The Truth.

 

Also, for simplicity -- and in spite of the fact that I usually prefer the modular approach -- I do not want to add a separate preamp anymore. 

 

Any idea whether the Holo preamplification module would bring benefits compared to the MiniDSP's attenuation?