Upsampling to anything other than your DAC's internal conversion rate

[Miska]

If your DAC chip internally converts musical input at DSD512 or at PCM384, should you be upsampling to anything other than that and if so, aren't you just adding more conversion artificats?

 

I'm currently upconverting everything to DSD128 in HQPlayer because that is the highest DSD rate my Oppo BDP105D accepts. To me, it sounds great. But am I fooling myself?

 

AFAIK, Oppo uses ESS Sabre. With Sabre chips, the DSD path through the DAC is much simpler and more straightforward than the PCM path. So you can happily send it any DSD. Regardless of the DSD rate it behaves the same way, but higher rates just improve performance.

 

Even if a DAC supports only up to 96k PCM, it is still worth upsampling RedBook to 88.2/96k before sending to the DAC. The first step up is sonically most critical. And doing 2x conversion would already put 22.05 kHz wide space between DAC's filter and the original content, so the DACs built-in filter is already having much lesser impact. In addition, with typical DAC chip, every doubling of input rate drops out one cascade digital filter section from the built-in filter chain.

 

In addition, many devices doing digital room correction or other DSP (AVRs and such) typically resample everything with no so great filters to either 96k or 192k to apply the DSP at fixed rate. Sending these already good quality resampled data will typically also improve performance.

 

Sometimes it may be useful to choose between PCM and DSD output though. In particular with Chord DACs it is better to use highest PCM rate, since they convert DSD inputs to that same one anyway it is better to send PCM out.

 

So no worries...

[Miska]

I assume that means they are also applying their own filters at each step? So every time I substitute a preferred HQP filter for the DAC Mfg's filter, I benefit at each doubling?

 

Yes. From this perspective, less you have filters left at the DAC side, the better. In many cases if you can reach 352.8/384k there are no filters left. That is also sort of down side, and that's why in many cases DSD is better option.

 

My other request, not of you, but of the community here, was to begin putting together a list we might all share as to what the "optimal" or "native" input rates are for at least the most popular DACs. Do you think that would be a worthwhile exercise for CA users in order to help newer HQP customers maximize their benefit from upsampling?

 

Yes, although the easiest way may be to list "exceptions to the rule". Like; "with Chord DACs, use highest PCM rate available".

[Miska]

I would disagree about Chord Dac's sounding better by upsampling PCM. From my experience with the Hugo this is not the case. Best to feed the Hugo with given bit perfect feed. In fact, I think Redbook sounds better streamed as is, not upsampled.

 

I don't disagree with upsampling with HQP in general for most DAC's, just not the Chords.

 

I only have Mojo, have not tried Hugo. But Mojo clearly performs best when it's volume control is disabled and it is fed with 705.6/768k PCM data. Volume can then be controlled for example form HQPlayer as necessary.

 

Of course if you prefer the sound/performance of the Chord's WTA filter, then you prefer it and there's nothing to argue. The most similar sonic signature you get from HQPlayer with poly-sinc-hb or closed-form filters.

 

For me, the second biggest improvement with Mojo was to disable it's volume control and use external headphone amp (Schiit) instead.

[Miska]

but the question that newbies and others here like myself want explained is how does upsampling add any enhancement to music from its native format?

 

The native format is just essential set of "cues" needed to reconstruct the analog waveform. It is a bit like kids "connect the dots" drawing, where you have numbered points and need to draw a curve through the points to get to a full picture. If you add more very carefully calculated dots between the original ones, drawing the curves becomes easier and more accurate. Analog filter at the DAC output is the one drawing the final curves, everything is just dots before that.

 

You can't add detail that's not there

 

It is not about adding detail that doesn't exist, it is about making it easier for the analog conversion section for to accurately reconstruct the waveform. Data is not the signal, the signal needs to be built from the data. If there is more data, it helps getting an accurate end result.

 

and if we are talking about ultra-high frequency noise or artifacts, who can hear that, anyway?

 

Through intermodulation mostly. If you have two high frequency artifacts, at 100 kHz and 101 kHz -> you get 101 - 100 = 1 kHz difference tone which falls into audible range.

 

Is there inherent distortion with PCM at Nyquist frequencies when played "straight"?

 

Yes, this is best demonstrated with a NOS DAC.

 

Let's first take a look at 1 kHz tone played through a DAC without any upsampling, you can see that there's 1 kHz tone around every multiple of the 44.1 kHz sampling rate up to over 1 MHz frequencies. These are called "images" that are supposed to be removed by the analog filter at DAC's output. Those are essentially distortion - jaggedness in the output waveform.

 

Then the same source data, but now upsampled to 384 kHz sampling rate, you can now see that there's anymore tone left repeating around multiples of the 352 kHz rate and much lower in level.

 

This is particularly clearly visible when you make a 0 - 22.05 kHz frequency sweep.

 

Here at the original 44.1 kHz sampling rate, you can see the entire sweep spectrum repeating around every multiple of the 44.1 kHz sampling rate, the lowest frequencies being the strongest ones (since the DAC's output low-pass filter has not yet cut into those as much).

 

And here the same source upsampled to 384 kHz sampling rate, now you can see much lower level images which are repeating around multiples of 384 kHz and as such the number of those is also lower (four pairs, so total eight).

 

Since these components are fully correlated with the source signal, also the intermodulation products are fully correlated.

 

I would LOVE for someone to suggest some specific Audirvana or HQPlayer PCM upsampling setting for my particular DAC that will enhance the SQ.

 

Unfortunately I don't have specific good values for the NAD since I don't have it and it uses it's own approach. I would just feed it with 176.4/192 kHz PCM data and try different filters. If that doesn't sound good, then hires recordings at the same rate are likely not going to sound great either.

 

But it certainly has it's own set of digital filters / upsampling.

[Miska]

But if one was inclined to go down the HQP route (it fulfills your format needs), then one should start with the proper DAC and streamer to get the most out of HQP benefits.

 

Thus back to the intent of this thread. Which ones?

 

Well, I though this wasn't about HQPlayer or any other player in first place, but more about finding which particular input format makes best out of the DAC. There are even cases where for example 44.1k and 48k base rates have different jitter performance and the other performs better. Many DACs have one certain "sweet spot" - one input format that gives the best performance.

 

Frequently there are are also settings on the DAC that are involved in the end result.

 

For HQPlayer, there are bunch of good DACs. Just some examples being T+A DAC8 DSD, iFi DACs, TEAC UD-501/UD-503/NT-503, exaSound, Fostex, Sony, etc.

[Miska]

Unless using a DAC that's purpose designed for such use or with upsampling that can be turned off, can we be certain that any software upsampling/filtering is not making things worse than better? How do you know that by software upsampling to the DAC's max rate you are somehow bypassing the DACs own processing - maybe its just doing the job twice and making things worse as a result?

 

If the DAC is based on a DAC chip from the typical set of manufacturers, then it is usually pretty clear cut. If the DAC uses chips from TI/BB, AD, CL (incl Wolfson), AKM or ESS for example, then the behavior is known.

 

I have a collection of DACs that keeps growing that I measure and test.

 

If upsampling makes things worse, then also playing equivalent hires content would be making things worse. Pretty simple. I wouldn't buy such a DAC in first place.

[Miska]

Miska - over on the Twisted Pear forum on DIYAudio I asked about this a few weeks ago, specifically what is the highest DSD rate supported by the ESS chip (ES9018) but really didn't get an answer. Do you know? In line with this thread topic, if I am feeding my DAC the highest rate possible then the less processing required by the DAC, supposedly a good thing.

 

I should be reading it more frequently than I am...

 

Note! I don't know details about the latest series ESS has put out! exaSound is first I've seen to use a chip from that series.

 

Based on the datasheet it should work up to DSD512 and there are some implementations. But it doesn't seem to be reliable.

 

The only requirement they have is that MCLK must be at least 3x the DSD BCLK and MCLK must be <= 100 MHz. I personally think it should work if one would use exactly the 3x clock. This is a bit hard of course with normal audio clocks. But if someone wants to experiment, someone could come up with some frequency like 90 MHz for MCLK and then I could add 30 MHz output to HQPlayer. That would of course require a bit of customization to the USB firmware too.

 

Currently, I am using HQP to send DSD256 data to my DAC, the limit being the JLSounds USB board supporting native (non-DoP) DSD256 on Linux.

 

That is, IMO, pretty good option.

[Miska]

Another DAC to list as "forget DSD and use PCM at 352.8/384k" is the new Prism Callia.

[Miska]

Upsampling to Dsd is a negative using the Yggy.

 

It colors the sound too much

 

Sure, because it doesn't support DSD in first place at all...

[Miska]

Considering the amount of filtering available these days, there is no chance in Hades of selecting the right filter 'to make the sound better' when rarely does anybody (I'm talking about the DAC user) know what the topology is in the DAC to begin with, let alone to apply an external filter characteristic or even consider increasing the sample rate.

 

I don't recall seeing, oh, the DAC has ESS9018 and 5th order butterworth filters and the optimum rate for this DAC is 2x upsampling, plus Izotope a,b,c,d, e settings, or polysinc..with dither..? This is all too hard!

 

It doesn't matter... The filter choice depends on the source material and preferences of the listener, not on the DAC.

 

Dither algorithms for PCM are up to your preferences, if you don't want to bother, just select TPDF and you are fine.

 

For modulator, it is also largely preference based, but for simple lowpass filter DSD DACs (Lampizator) it may be technically better to choose lower order modulator (5th order) and for the ones with more complex lowpass filter (DAC chip based and T+A) it is better to choose higher order modulator (7th order).

 

After changing to A+ a month ago after about 18 months on HQPlayer, the sound for me is better with no upsampling, leaving the filtering bog standard and play the file in its native format, whether 96, 192, DSD64 whatever.

 

The DAC cannot know if for example 192 kHz data is result of upsampling or happens to be hires source originally created at 192 kHz. For the DAC it is just data sampled at 192k.

 

So now I just accept what the DAC designer does and work on the sound that goes in and comes out.

 

You could also accept my defaults or recommendations and stick to those without playing with the settings. This is no different from sticking to DAC designer's choices. Noting here that "DAC designer" is usually the engineer at the company who manufactures the DAC chip and he's under strict resource and thermal budget, so has to make many compromises on his design so that it

1) doesn't increase the chip price above the set price point

2) as a result of (1) can fit within the computational resource constraints of the chip

3) doesn't exceed set power consumption limits - meaning that the DAC chip cannot require heat sink or active cooling

4) fits in the chip manufacturing process

And once the design is set, it is manufactured at least for 10 or more likely for 20+ years without changes or possibility for updates/improvements. The improvements then come as a new product at some point.

[Miska]

So if I understand correctly there are DSD-able chips and those which do PCM only.

 

Yes, and for the TI/BB models the PCM path is pretty much the same with some differences in digital filters and converter output stages. But the overall architecture is the same. They've omitted DSD capability from the parts that are pin-controlled ("H/W control" in their terminology) instead of I2C/SPI software control. This is because the number of pins that can be used for controlling the DAC is limited. With software control over communication bus (I2C and SPI) there are no such limitations.

 

There are two otherwise identical models, PCM1794A and PCM1792A, where 1794 is pin-controlled and 1792 is software controlled. And 1792 supports DSD.

 

The latest flagship model PCM1795 also supports both PCM and DSD.

 

Some manufacturers don't want to bother to have a microcontroller (and the associated programming needed) in the DAC and want to use easier pin-control DAC chip models. In the past I wrote the necessary control software for various S/PDIF receivers and DAC chips for a small microcontroller. It took a week or so to get everything done and working, but I have strong software development background. The MCU (microcontroller) itself costs just $1/piece, so it is not so much of a cost issue.

[Miska]

Which commercial DACs besides the Teac 501 can have filtering and possibly the ASRC turned off?

 

You mean digital filtering? Analog output has analog filter as it should...

 

iFi, TEAC -503 models, T+A DAC8 DSD, hiFace DAC, etc... Metrum DAC's don't even have any digital filters (PCM NOS DACs).

 

Chord Mojo also when you run it at 16x input rates, I don't count the following linear interpolation (argh) as a filter.

 

Many also have adjustable analog filter.

[Miska]

From my listening there's a category missing - best with bit perfect input, DSD or PCM . I'd definitely put both the Hugo and my current Esoteric in that category -noting that the latter is a very close relation of the TEAC dacs you mention above.

 

Chord also performs better when DSD is converted to PCM before sending there, because they have less than great DSD to PCM converter inside.

 

How do you know how your "bitperfect DSD" you are sending to your DAC came to be? (or PCM for that matter)

 

Chord Mojo definitely goes into category "disable it's volume control and feed it 16x 32-bit PCM".

[Miska]

If I understand correctly, upsampling affects the audibility of filtering more than anything.

Current knowledge states that 2fs puts the filtering too close to the top end of the audible band and because filtering requires some slope there will be audible consequences.

 

My question is when we upsample, shouldn't the filtering become inaudible?

I thought that one would be filtering (the upsampled redbook) at 4fs or 8fs, etc.

 

There are two things involved, digital filtering and analog filtering. Digital filtering exists to help analog filtering, which is always mandatory, to do it's job better. Reason is that it is feasible to create accurate steep filters in digital domain and avoid extra problems of filters operating in analog domain (noise, distortion, etc). This way the analog filter corner frequency can be moved away from the audio band and the slope (filter order) can be relaxed.

 

If you start with RedBook, there's no way you can avoid having the filtering (or lack of) implications right at the edge of the audio band. You just need to pick something that is as optimal as it can be. I've spent enormous amount of time to come up with filter design methods that are as optimal as possible in both time and frequency domains simultaneously. That is tough challenge to push closer to the limits, because ultimately the two are mathematically bound by the 1/x relationship and the Nyquist fs/2 limit puts hard boundary for the bandwidth. I take objective-subjective approach to my work, so I want things to measure well and once they measure well they also need to sound good. Otherwise I'm not happy and wouldn't have peace of mind. But ultimately you need to make your own choice based on your particular hearing sensitivities and the material you listen. That's why there are all those options with linear- and minimum-phase responses and such. Different types of content emphasize different kind of filter properties, so it is good idea to select a filter that fits both your hearing (different people are particularly sensitive to different sonic properties) and your material.

 

The (anti-alias) filter properties become embedded to the source material when the sampling rate is reduced to 44.1k. With apodizing digital filters it is possible to replace/modify these filter properties that have been embedded to the source material. With suitable apodizing filter you can get "de-blur" effect.

 

For example rock recorded in studio puts demands on the transient response, so a minimum phase filter that doesn't have pre-ringing is usually good fit, since there's usually not much real acoustics involved anyway. While classical music recorded in real acoustics puts demands to the sound field/space, so a linear phase filter is usually good fit, since there are no strong transients or at least very few.

 

As the source sampling rate goes up to higher resolutions, the effect of filters also gradually diminish. So for hires content you have less to think about in that respect. It is much easier to do good D/A conversion for hires than it is for RedBook! For delta-sigma DACs the effect of modulator however is persistent at the same level regardless of the source sampling rate. And that is the another 50% of the DSP-side performance involved.

[Miska]

The DAC may not understand where the 192 came from, but 'we' know the file has undergone sample rate conversion.

 

It has gone through it anyway...

 

There maybe more processes out in the wild but I only know of Izotope that's built into Sound Forge and Audirvana +. The Izotope SRC can be change the steepness, max filter length, cutoff scaling, alias suppression and Pre-Ring. I don't have much of a clue how to set them (the point of another thread), let alone make conclusions how they work. On the occasion I do SRC to create a CD from a 96/24 file, there's nothing much lost, so by luck SRC works.

 

Sure, you can find comparisons for some of those here:

SRC Comparisons

 

The ones listed under title "Signalyst 2.9.1" still apply to current versions of HQPlayer.

 

I don't offer iZotope kind of adjustments, because those are not enough to describe all the filter properties, just a minor subset of it. And still they offer a way to shoot yourself in a foot.

 

By inference, you're telling me that upsampling SRC is transparent? If you apply mathematics, no matter how 64bit precision you make it, there will be a difference.

 

No, I'm telling that there's no way you can avoid upsampling SRC with any of the modern DACs. Either you do it in software before sending it to DAC (and DAC just thinks it's a hires recording). Or you let DAC do it. The DSP engines inside DAC chips are nowhere close in accuracy and quality to what you can do with software. In modern parts those are typically 32-bit and in older parts 24-bit. And quality of the filters is also severely limited due to the issues I said.

 

Another item is quality of the delta-sigma modulator in the DAC chip.

 

Overall, most DAC chips only run upsampling digital filters to 352.8/384k rate and then go up from there by copying samples or by using linear interpolation (AKA the dumb old mathematical mean). While for example HQPlayer can upsample with digital filter right to 24.576 MHz rate and doesn't ever use cheap methods like copying samples or linear interpolation to increase rate.

 

Some examples...

 

Here's output of 1 kHz -60 dBFS sine with 44.1/24 input data, you can see the early increasing modulator noise floor due to 3rd order modulator and some idle tones too:

 

And there's output of the 1 kHz -60 dBFS sine upsampled to DSD256, you can see that noise floor is now flat and the only extra peaks left are the low level noise peaks from the XMOS USB receiver (USB packet noise):

 

...of course you could still subjectively prefer the first one...

 

Or from another angle, 0 - 22.05 kHz sweep input at 44.1/24, standard digital filter selected, you can see images around multiples of 352.8 kHz:

 

Same source data, now upsampled to DSD512, no images at all because it has gone through proper digital filters to 24.576 MHz rate:

 

...of course you could still subjectively prefer the first one...

 

 

Sure it takes much more processing power to upsample 256x or 512x compared to 8x done inside the DAC chip. Especially because the precision is also immensely higher.

 

Others have posted that they prefer not to upsample and I would agree after listening to various sample rates for HQPlayer, including DSD256 for 18 months. I don't believe it is problem that the computer is working harder and is noisier as a result, it's the FLAC (uncompressing) vs WAV argument, no. Maybe Izotope is better.

 

I cannot know why you prefer something and cannot argue about it. I can just argue about objective things.

 

Maybe I drag out the TASCAM DA-3000 recorder and make some comparisons. Now, which filter to use?

 

I'm not sure what you are talking about. Your filter selection should be based on the material you listen and your personal preferences.

[Miska]

An example I've used before is that my main system speakers are set up (including the crossover filters) to be time and phase correct, which may be a reason I prefer linear phase to minimum phase filters there. It seems to me subjectively that this results in better imaging (and in fact I've done a bit of blind testing where I've preferred linear phase).

 

As I've been trying to say before, this is one of the main decision driving factors among type of the content. I listen mostly through headphones, and mostly through Sennheiser HD800. Since it is single driver one-way solution like most headphones, it is time/phase coherent. But since I listen a lot things like prog rock (mostly, but really everything from classical to electronic club music), there's not much "imaging" to talk about in that genre (IMO), but I'm particularly sensitive how the drum/cymbal highs sound like and how their leading edge sounds like. Overall, I'm most bothered by problems at the top end of the audio band, like resonances from metal dome tweeters. Any problems on that area, like hissy tweeters make me feel uncomfortable. Problems in that area can be emphasized by certain leaky digital filters. As a Finnish person I guess I'm supposed to like Finnish speakers like Genelec studio monitors, but I really cannot listen those. Damn, almost all commercial Finnish speakers use metal dome tweeters. I love ribbon tweeters though, fast and clean and wide. Silk domes are fine for me too, although less precise and wide. That's why I now use both Elac speakers with their JET tweeter (ribbon in AMT configuration) and Dynaudio with their silk dome.

 

But that's me and I think I know how to switch my listening focus to other things and I know for certain that other people are sensitive to other things and listen different types of music. So there are various different options with emphasis on different things. I go to classical concerts too, especially because I like the new Music Centre at Helsinki.

 

My favorite album is Pink Floyd's Meddle. The highest resolution versions of it I have are all RedBook (couple of versions). But damn I have tried hard to make best out of it, and I believe I've got pretty far, IMO...

 

choices "apodizing" filters, to substitute the better transient performance of the HQPlayer filters for worse transient performance that is often characteristic of the filters used in the creation of CDs. On the other hand, there are people who don't particularly like apodizing filters, which leads to what I say below.

 

There are still both number of linear- and minimum-phase versions of the apodizing filters to choose from. But there are alternatives of course too, like poly-sinc-hb and closed-form that are non-apodizing and from practical perspective don't change anything in good or bad.

[Miska]

Actually' date=' nearly all [i']finished DAC products[/i] for sale are superfluous to your preference of upconverting to DSD256 ? In the sense that you'd rather have no further processing, just a low-pass filter ?

If so, what good are finished DAC products ? What audience, for what purposes are they marketed towards ?

 

I'm not sure I understand your question. They try to provide single-box solution manufactured to certain price point making certain amount of profit.

 

From my perspective, they have features that are not necessarily needed. But if one doesn't have, or doesn't want to have, external processing entity, they can be self-contained too. But most DACs that are DSD capable are based on COTS DAC-chips. For example the AKM's AK4490. It's 2.5€/piece, if it happens to have some PCM side to do rudimentary digital filters and modulation, it doesn't matter too much in the big picture (and doesn't do harm) as long as the DSD conversion is good quality. If someone want's to use the on-chip DSP engine needed with PCM-inputs, I don't really care.

 

But I'd rather have DAC manufacturers spending the money to make the actual digital-to-analog conversion as good as possible and leave the DSP processing to others. For example the T+A DAC8 DSD is good example of such for the DSD-side. It would be even better if they'd leave out the (to me unnecessary) PCM side out altogether and use the money to improve the DSD side.

 

The only thing that I feel comes close to what I want is dCS with their upsampler + DAC device combo. But the price is quite high and I feel I get quite a bit better performance for a lot cheaper now. So I wouldn't buy one even if I'd win in a lottery.

 

 

P.S. I have for example TEAC NT-503, it has lot of features, but the only thing I use from it is DSD256 input over USB. Everything else is unnecessary extra, but I don't mind since it doesn't interfere with the stuff I want to do. (the DSD upsampling they have built-in is horrible quality and nobody should use it)

[Miska]

The selection of the filter is still a crap shoot with some really deep understanding of which filter works at its optimum with:

 

a) the music content, eg complex orchestral, jazz trio, or compressed rock

 

For that, I have the recommendations. But in any case you can just try out or make an educated guess based on the recommendations.

 

b) the type of DAC chip used.

 

That's about modulators and not about upsampling filters. But I have tested/measured my modulators with different DAC chips, and designed those to work well, that's my part of the job. And for the discrete (non-chip) DSD solutions I have the recommendations on what to use.

 

You're all over it, I'm not, the more I look at filter selection, gives me the heebee jeebees, where's that crucifix. It begs the question then to introduce profiles for different types of music, gosh even down to the album (?), so the filter has an optimum, or better chance of working.

 

Doing these things with software is possible. While DAC manufacturers try to pretend that such doesn't even exist, because the DAC cannot know what genre the music is about. All you get is either "one size fits all", or couple of filter options to choose from. Unfortunately they are usually quiet about how you should be selecting between those filters. While I at least try to formulate a logic on how (IMO) the filter selection could be done.

 

Even order harmonic distortion is a good thing then, music has gobs of harmonics, so makes sense to add it in, rather than being clinically cold.... any plans to add it as a "filter"? A+ does, I believe it's Mode 2.

 

No, I don't like those kind of ideas. If the recording is clinically cold, it should sound like clinically cold. But the old fashioned digital stuff used to make even warm sounding recordings sound "digital", "clinically cold" or "hash in the HF" due to those HF artifacts. It's like some of my first CD players with SAA7220 digital filter and TDA1541A DAC chip - making everything sound digital - the sound of 80's. Like the plastic Casio keyboards.

 

Computer audio does make selection of music very easy, rather than shuffling through discs, but to get the optimum out of it, is a lot of work hardware wise and software wise to get it sound less digital. Makes me wonder and why bother with it all.

 

Primary problem is RedBook content. Some engineers really wanted to push storage requirements to bare minimum in 70/80's and really followed the theory to the extremes. Making it very hard to properly go back to analog. If you'd have content in 192/24 PCM or DSD you'd have much less problem to begin with.

 

Luckily they didn't have processing power to do psycho-acoustically lossy codecs at that point (in a commercially feasible way for the playback side)!

[Miska]

By horrible DSD upsampling, I take it you mean that the filter on its chip (the AK4490) to lowpass DSD input (for volume adjustment and remodulation) is low quality.

 

No, I mean the DAC has it's own separate PCM-to-DSD converter you can enable/disable at will. I'm not sure if it's made using AK4137EQ chip or if they have their own implementation in DSP/FPGA.

 

The 0 - 22.05 kHz sweep looks like this with the built-int PCM-to-DSD conversion enabled and 44.1/24 source:

 

With DSD inputs the DAC behaves just fine. With PCM inputs (using the on-chip digital filters) it behaves like most other DACs, with images around multiples of 352.8/384k because the digital filter can do max 8x.

[Miska]

Start by recounting your personal experience with whatever specific products you esteem and we can examine if it can be applied' date=' adopted by others.[/quote']

 

We've discussed quite a number of products before. Like especially the T+A DAC8 DSD, iFi iDSD series, TEAC UD-501/UD-503/NT-503, exaSound DACs and Mytek DACs. And also others like Marantz, Resonessence Labs, Sony and Lampizator DACs. With those, DSD works great.

 

And then there are category of DACs where DSD is sort of primary function, like EMM Labs/Meitner, Playback Designs and dCS.

 

Then there are DACs where it is not so great like Chord DACs and with those you are better going with the highest supported PCM rate.

 

But overall, as long as the DAC uses a DAC chip, the amount the DAC device manufacturer can do about it is pretty limited. Both good and bad.

[Miska]

For example' date=' youself can give more substance of your operation of a TEAC NT-503, detailing not least equipment chain (which isn't listed on your Profile) and listening-room layout, plus what music sourced, fed.[/quote']

 

I have so much equipment in different configurations that I don't see any point in listing all those.

 

You gotta proof better than simply saying :

 

I've been posting quite a bit of my own measurement data and such. But I have not seen such from you. Have you made any measurements that proof the contrary?

 

Are you trying to say that the equipment I listed works bad in DSD? I'm pretty sure everybody would be happy to know what is the input format that works best with those.

[Miska]

Does your measurements really prove how you hear ?

 

As I've said before' date=' measurements are a good starting point, giving objective view. Everybody needs to listen on their own to make their own subjective opinion.

 

I already described above my objective-subjective approach and not going to repeat it anymore.

 

Here's your chance today to show us the analogue eventuality—your preferred equipment setup and listening-room environment—from your digital mastery.

 

I've said already couple of posts above, that my primary means of listening evaluation is through Sennheiser HD800 headphones. I have other headphones too (AKG, Beyerdynamic and Stax), but those Sennheisers are my primary ones at the moment. The current set is both extremely neutral and at the same time very accurate and analytic, for pointing out any sonic problems.

 

I was already talking about my loudspeaker preferences earlier in this thread, if that matters to someone.

 

So far, it's been a leap of faith that it translates well in widely applicable, real world-listening-situation terms.

 

My subjective listening opinions are mine and I encourage everybody to form their own ones. My subjective experiences apply only to me.

 

 

P.S. Of course, when using loudspeakers, important part is digital room correction. One of the reasons for me doing lot of the development, because I wanted to apply digital room correction also to DSD recordings without intermediate PCM decimation.

[Miska]

Many readers are not new Hi-Fi enthusiasts' date=' they can spot troubling anomalies if you show them a photo of any setup practice.[/quote']

 

Why do you care about my setup? Whatever my setups are, they don't mean anything to anybody else since I'm the only one listening through those.

 

Well, for people interested in your filters, will they subjectively experience the same sound as you ?

 

I'm pretty sure not. You cannot find two people who would experience same thing same way. That's why people need to listen and make their own mind. Trial is free, so trying it out doesn't cost anything but little bit of time and effort. If you don't like it, you don't buy it. Pretty simple.

 

I don't see other companies regularly posting details of their listening rooms either. Maybe some do, but I don't really care.

[Miska]

It's mostly true' date=' purchasing Hi-Fi, we've often, always on-our-own :[/quote']

 

That's how it should be.

 

Just take tips online what to hear yourself next.

[Miska]

Miska, have you tried DSD256+ into a SET Tube Amp?

 

I don't have at SET tube amp, but I have a Prima Luna with my favorite selection of tubes. (2x Tungsol 12AX7, 2x JJ ECC802S, 4x SED EL34)