If I wanted something high-end, I would go external, and avoid any electromagnetic interference inside the PC entirely, instead of hoping some vendors EMI shield actually works decently.
Hhmm, I'm sure they have some experience as audiophiles, but that doesn't stop them from taking the "bigger numbers are better game" too far.
Let's take their dynamic range for example. 24-bits are enough to encode 140dB of dynamic range, and arguably even 16-bits with dithering reaches 120dB of dynamic range. And then their hardware only supports 123dB of dynamic range! That 32bit audio support is just pretty misleading marketing then, if your hardware can't output a full 24bit by a wide margin.
But at least 32bit audio is not *worse* than 24 or 16. It's just not better. The really dishonest trend is those 192kHz and 384kHz sample rates. Counter-intuitively, higher sample rates is NOT something you want in digital audio =)
The short of it is that 44.1kHz and 48kHz are legitimately enough to perfectly (per the awesome Sampling theorem!) encode all sounds your hears can perceive, anything higher is ultrasonics that you can't hear, but that can and does add noise over the whole spectrum. That audio is there and has to be processed when you chose a high sample rate, the problem is that audio hardware can't low-noise over the whole spectrum, not at any reasonable price.
Depending on how much your audio hardware prioritizes quality in those ultrasonic frequencies you can't hear (and that's unheard of, pun intended!) the noise might go from imperceptible to shockingly awful.
All in all the card might be good, the THD+N value looks totally acceptable for a PCIe card. But the truth is it won't be better quality than a cheap USB DAC. If they know what they're doing like you say, then this card's trying to take advantage of audiophiles who don't understand all those numbers, and I can't recommend supporting this.
>Counter-intuitively, higher sample rates is NOT something you want in digital audio
Well, this isn't really accurate. Higher sample rates are useful because they loosen the requirements of the analog filters in the DACs and ADCs, which are limiting factors. There is no audio information above 40 kSamp/s but if you want those dynamic range and SNR numbers then you need the higher sample rates.
I won't say there's no justification whatsoever for higher sample rates. Some margin is useful in analog filters (but again within reason, or any non-linearity near high-frequencies in your filters will further degrade the lower part of the spectrum).
But at the end of the day I can't take those numbers at face value and tell people that this is a superior product. I'm happy to claim that card's 384kHz/32bit output versus a similarly priced USB DAC would win in neither listening tests would nor objective noise metrics in the audible range.
The majority of DACs will internally oversample the input for filtering before analog conversion. Due to those attributes of most DACs, a higher input samplerate is not required.
And how good are those oversampling filters? I'll grant you than 384 kHz is overkill, but if the source material (e.g. mastering software, digital synths, etc.) can produce real signal at that rate, then oversampling can only be worse (i.e. by introducing artifacts).
And for source material not at 384 kHz, your software can use whatever oversampling technique it wants, rather than being at the mercy of whatever the DAC does inside its little black box.
There's no reason to have source files above 48 kSamp/s. The extra 8 kSamp/s covers the non-linearities in the ADCs. Since all of the data you want to reproduce is 20 kHz and below there's nothing to be gained from higher frequencies. Upsampling is a very simple process and the minutiae of the interpolation filter doesn't really matter. It's all simple digital logic running at painfully low clock rates.
So, you're arguing against oversampling DACs? I think that argument was lost over 30 years ago.
While not as common as 44.1 or 48 kHz, 96 and 192 kHz content has been with us since the mid 1990's. Since then, most recordings have probably been mastered in 24-bit/96 kHz or above, providing a massive catalog of potential content. These days, you don't need a new distribution format, like DVD Audio - you can simply add the format to new or existing streaming services.
And if you think the minutiae of the oversampling filters doesn't matter, then you're no audiophile. But, I think we've established that.
If you'd read the other comments you'll see that I'm not arguing against oversampling DACs. I'm arguing against oversampled content.
Also throwing labels around isn't something I'm interested in, but since you've opened that up: no one should be proud to be an audiophile. Audiophile is a label that's been hijacked by a delusional group who throw objective measurements out the window because they don't want to put the effort in to understand what's going on. They're happy to be fooled and be taken for a ride. You're right. I am no audiophile.
The minutiae of the interpolation filters don't matter in that their audibility is incredibly low and as long as you're not picking something silly like NN or linear you won't see large performance differences. The differences will be there and you can go as far as picking neural net based algorithms, but you won't be able to hear the difference.
Sure, there's a lot of stupidity in the audiophile community. I don't defend that. However, what you're missing is the ethos that one can only achieve the best audible performance, from an entire system, by optimizing each stage *beyond* the audible range. By definition, this can only be done through engineering, and measurement.
And your assertion that the interpolation filter doesn't matter only applies to cases where the transition band is *completely* above the audible range (i.e. content that's already oversampled).
Finally, everyone here is forgetting that Shannon's Sampling Theorem only applies to *periodic* signals. Unless you sit around and listen to sine waves (or other repeating waveforms), audio data is not truly periodic. In other words, sampling theory is merely an approximation, for audio. Oversampling at (or close to) the source gives you additional margin for this, as well.
So, if you're trying to avoid over-engineering (and let's face it - the extra data we're talking about is cheap, by modern standards) and are satisfied with something that's merely decent, then good for you. Please just don't force your values on others.
@willis936 You are absolutely right. higher sample rates are not for the bits you can't hear, it is to make things as analog as possible before it hits the analog hardware. So, lets say (for sake of argument) that -1V is the bottom of a wave form, and 1V is the top of a wave form, and we have some 440 'waves per second' for a C note. Divide that up 44100 times and you end up with a whole series of voltage readings which can be used to reproduce that note. In digital, that is fine to reproduce something to the human ear, but to analog hardware (and especially for good hardware) this is a very long time between samples. The hardware can over-shoot, or under-shoot the next voltage reading which leads to weird jagged stair-stepping where it should be a nice even curve, which introduces weird artifacts in the music. Up-sample that to 96-192Khz (beyond that is perhaps overkill), and now you have lots of voltages for the dac to read, and far less room for audible errors to be introduced by literally bored hardware that is just waiting to see what is next. Its not about creating tones that you cannot hear (but give dogs a more satisfying experience), it is about transitioning the stuff you can.
Caeden, that's not right. All you really need to think about is the #bits (which limits your DR), sample frequency (which limits your frequency response), and the filter at the end of the chain. When you start thinking in terms like "oh the stair step is too steppy", then you're not understanding how the filters work.
No, @CaedenV is exactly on the right track. In the 1980's and 1990's there seemed to be a race to reach ever-higher levels of oversampling, in DACs and CD players. I'm pretty sure I remember even reading about 256x oversampling, but 8x was not uncommon. I have a DAC from the late 90's with a Pacific Microsonics HDCD interface chip that used 8x oversampling.
At a core level, the output voltage of PCM DACs is latched. This introduces "stair steps", like those having been noted. By increasing the sample rate, the noise introduced by this process is reduced in amplitude and pushed far outside the audible range, making it more easily addressed simple analog filters at the output.
@mode_13h, you're wrong. Because analog filtering is on both sides (on the input for recording, on the output for playback), the 'stair step' you talk about doesn't ever actually exist. Yes, oversampling does technically allow you to use a crappier analog filter, but modern technology doesn't have this issue unless it's designed poorly. Once filtering is applied, the frequency will be accurately represented. This is just fundamentals of Nyquist-Shannon sampling theorem (https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shan...
Watch this video: https://xiph.org/video/vid2.shtml - if you care about 'stair stepping', skip to chapter 3 specifically which addresses that. The creator of that video goes into why sampling rate doesn't matter beyond say, 48 kHz.
*sigh* this is kid stuff. Covered in the first chapter of any signal processing text.
Yes, you *do* need to band-limit the signal, before sampling, on the upstream side. No argument there. Failure to do that will result in aliasing, which is impossible to remove, unless you can make some assumptions about the signal (in which case you're also not actually using the full bandwidth of the channel).
Second, he's glossing over what the device is doing to groom the output. His test would be relevant to my point if he had a PCM DAC chip on a breadboard - not testing an end product.
Finally, see my above point about sampling theorem vs. audio. He's using sine waves, since they're easy to understand and work with. How one can actually quantify the performance of a system is by passing through a full-bandwidth signal and subtract it off (or divide it by) the original. That's how you'd be able to observe any corners being cut along the way.
Are we not talking about complete systems here? I'm sure if you use some raw DAC with no filtering then weird stuff can happen. No question there. Then you aren't properly band-limiting a system.
Your above post talking about sampling theorem vs. audio is incorrect. Applicability is to periodic OR non-periodic signals. I can cram as much as I want within the bandwidth of a device and it should be able to be accurately reproduced. If I have a single step waveform sampled the only way to actually represent that is with ripples and a decay that would be in accordance with the sampling rate I have available. There's only one pattern of signals that would satisfy that waveform. Yes you have to worry about intermod distortion and other non-linear effects when you start throwing more and more signal content in there, nothing is perfect, but these issues are relatively minor.
What are you actually claiming is the benefit of oversampling DAC in relation to the output voltage? Closer accuracy to the signal because there's more points as it 'approximates' it? With a filter that doesn't matter.
I'm not sure how higher bit depth or higher sampling frequency fix any of the problems with 'corners being cut'. I'm sure I can design a crap 384kHz 32-bit DAC and an amazing 16-bit 48kHz DAC; the marketing numbers don't really have anything to do with how good it actually performs.
Regarding sampling, you're contradicting yourself. Shannon says you can't have aliasing if you stay below the Nyquist limit, but that's only for periodic waveforms. Now, you're trying to add a bunch of caveats and tell me that I can't *really* use the entire channel bandwidth? Just read the paper I linked and bring yourself up to date, circa 20 years ago.
The author also made presentation on the same subject material, 10 years after that:
And what I claim is the obvious - of course the quality of the interpolation filter matters! And no, your analog output filter is not magic. It can't make a non-oversampled signal/DAC perform as well, and it doesn't make the quality of the oversampling filter irrelevant.
I suppose you've never heard of noise shaping, either.
> I'm sure I can design a crap 384kHz 32-bit DAC and an amazing 16-bit 48kHz DAC; the marketing numbers don't really have anything to do with how good it actually performs.
It's cheaper and easier to get good performance with an oversampled signal. That was the whole point of delta/sigma DACs, but it also holds true of PCM DACs. You know enough to have opinions, but you're no DSP engineer.
As for bit-depth, it depends on the original signal. 16-bit doesn't quite match the dynamic range of human hearing. For home listening purposes, what's interesting with 24-bit (or more) is you can actually cut out the analog preamp and run your DAC straight into your power amp, doing your volume control in the digital domain.
I forgot to add that bit depth also affects phase accuracy. Of course, if your input is only 16-bit, then oversampling that obviously won't improve accuracy that's already been lost (though it can mitigate the impact of downstream filtering).
This brings up an interesting point, however. Because the high frequencies in music tend to be fairly low amplitude, their effective bit depth is lower. To address this CDs have a feature called pre-emphasis, which is essentially an EQ curve that boost high frequencies during mastering, and a flag telling the CD player to attenuate them after the DAC (although CD players with oversampling and/or higher native bit-depth DACs can do it in the digital domain).
I think we may actually be more in agreement than not in terms of actual hardware info - we're just approaching this from different lenses. I care about full systems here and what the marketing wank numbers actually mean to my audio quality. What the chips do internally is often times hidden from users. Using a delta-sig (that internally does over-sampling) to achieve a better bit depth as a result is fine by me. It's just a means to an end. My point being that the label of 24 bit or 384kHz sampling (or even 192kHz sampling) is definitively not a label that must mean 'good audio quality'. There's a ton of other factors at play. Usually the fact that my audio doesn't have a bunch of interference from nearby noise sources plays a bigger role than all this other nonsense.
That being said, any single audio DAC chip won't look like a stair step on the actual output due to inherent limitations of slew rate/high freq attenuation internally. Also, any audio chip you buy worth it's salt is going to have filtering already built into it. Most seem to quote at least 100dB of attenuation at ~0.45*Fs.
Also, not sure about your point on sampling? Where did I say you can't use the entire channel bandwidth? My only point is that non-periodic signals (to be 'truly' represented) effectively need infinite bandwidth to represent. Sure, after only a short period of time, the need for the extra bandwidth is lost in noise, but if we live in the perfect world (where DACs have nice sharp edges on their stair step output) then you need infinite bandwidth.
Also, it's definitely not cheaper to get good performance out of a higher bit depth DAC. Silicon costs money, full stop. If you can design a DAC that performs just fine with 16-bits and lower sampling rate, that is going to cost less. Unless you mean engineering investment to actually construct that device... then maybe there's a tradeoff.
Lastly, I think we'll have to agree to disagree on 16-bit vs. 24-bit usefulness for listening though. Same can be said of emphasis additions. There's plenty of tests that show people can't tell the difference between 16-bit and 24-bit well constructed audio.
I agree that sample rate and bit depth cannot be used as a simple proxy for the audio quality of a device.
Second, I figured most audio DACs have built-in analog filtering, for probably a long time now. Probably also oversampling, as well. That doesn't make my point invalid, as I'm concerned with the burden sample rate places on the design and performance of the analog output filter - whether or not it's integrated.
My point about periodic vs. non-periodic signals is that adding more bandwidth should allow more headroom to handle the transients other aspects where audio signals behave differently than truly periodic waveforms. I think 96 kHz should provide plenty of margin.
I never said that a DAC with a larger word length would be cheaper. Again, I'm concerned with sample rate. And to that end, the reason higher sample rates and oversampling enable better output quality per $ is due to simplifying the demands on the analog output filter. With something like 96 kHz, you can use an antialias filter with a wide transition band, simplifying the task of minimizing resonance and maintaining good phase linearity.
Most of my audio library is 44.1 @ 16-bit, and I enjoy it quite a bit. I think where you're likely to hear the limitations of that format is in the corner cases. Also, we don't know how much recording engineers are constrained by it, in production (though, only a concern for less "pop" recordings).
BTW, one of the infuriating things about comparing CDs to vinyl or DVD audio is that the mix is usually different. So, you're not really comparing just the media or format. But I never bought into the vinyl hype, and its copy protection kept me away from DVD audio.
They're just citing the figures of the DAC/ADC, and pretty much no actual products ever meet those specs, so don't read too much into that stuff. Even pro stuff doesn't typically come close to the rated specs of the DACs (some can, but most are probably a good 10-20dB below the rated spec of the DAC). If its 100dB or above its decent. Oh and I'm talking about the final signal post analog section (as its not difficult to get the rated spec out of just the digital realm).
Yes 32 bits is rather pointless and largely marketing fluff (it'd be useful in audio production and for archiving or something, but is largely pointless for consumers), just like the stupid high sample rates (last I saw they were up to 768kHz, and are at 22.xxx on DSD stuff). But that's just how audio is these days sadly. You should look into DSD vs PCM, and even Delta Sigma vs multi-bit (R2R for instance) and similar discussions raging in the audiophile world these days. They're desperate to keep pushing people to upgrade. There's one company that touted its "medical grade" multi-bit DAC chips that they then admitted (on forums) test much worse than even mid-range PCM DACs for audio (because those multi-bit DACs are not being used for audio in the medical field, and the medical equipment using it are expensive for reasons other than the "it meets the specs" DAC chips used), but still try to claim that its world beating audio quality. This company has a cult following. They make some decent stuff, but despite them claiming to be beyond audiophile nonsense, some of their products often don't meet basic electronics (not properly grounded, they had one that was unloading DC feedback when turned on, which was visibly distorting headphone speaker diaphragms; to be fair they did fix those issues in production on some of it, but somehow new products still come out with some of the same issues from time to time - and its not because "bad run" during production, its because they'd overlook stuff like that during the engineering phase).
I won't say it wouldn't be better (there are other factors, and yes, some like shielding from noise that will be to the USB DAC favor), as there's other issues. I've used a variety of external audio gear and keep going back to my X-Fi Titanium HD (which this card is pretty similar to, high quality ADC and DAC). It has some flaws to be sure (can definitely get some noise from other components from time to time although that's been rare, and sometimes due to Windows Updates it'll reset the software settings - which is discernible via listening - I turn off all the processing of the card and put it in Audio Production mode so that it just passes bits to the DAC and then analog out), but still sounds good to me. I've tried a variety of USB DACs and keep going back to just the Titanium HD. This included some popular with audiophile DSD and multi-bit DACs, and neither of them sounded right to me, and something audiophiles seem to want to refuse to accept is that both types of devices (DSD and "multi-bit") tend to do a lot of shaping and filtering of audio signal and that is what they're hearing (so its not "true to the actual audio signal" like some of them claim). And that they could accomplish the same with PCM (which PCM does some things of its own, and some audiophiles know that and just claim that they prefer the filtering of one over the other, or often the specific implementation of some piece of expensive audiophile DAC they bought), but they'll frown about EQ and other "processing" calling it fake and "digital sounding" and other fairly nonsensical phrasing. And some programs have been working to add some of the distortion that some people find appealing that is cited for vinyl and/or tube based "analog sound" that some try to claim is why they sound better.
Ironically, delta/sigma DACs originated as a lower-cost alternative to PCM. Then, I guess DSD is a consequence of some folks getting carried away with the idea.
The funny thing is that I'm pretty sure even DSD recordings are mastered (if not also recorded) using PCM-based software. So, you're doing at least one inexact conversion, which might not have audible artifacts but I'm sure it has some.
> The really dishonest trend is those 192kHz and 384kHz sample rates. Counter-intuitively, higher sample rates is NOT something you want in digital audio =)
Where are you getting that? Sounds a lot like an invented fact. Why do you even care about low noise, in ultrasonic frequency bands? You won't hear it, by definition, so it's no matter if a little is there.
As said by others, a higher filter rate improves the effective performance of low-pass antialiasing filters. You can put the entire transition band above the audible range, along with any resonance and leakage. Filter design is a game of compromises. Oversampling eases the burden, substantially.
I'll grant you that 384 kHz at the input is overkill, but DACs have long oversampled to such frequencies and beyond.
Let's just say how 96000Hz at 24 bits is just perfect enough. You need a higher range than you can here so that you can parametric adjust the high range that you can hear but that it doesn't interfere with lower high range & mids. Still real magic lies in getting the range from 128 to 600 Hz (vocals) right & to push the 12~19 Hz range (sub bass) high enough to be auditable. SNR is single most important metric along with actual dynamics (dB) range that makes what we call sound stage and separation. Still analog end listening device is the one most important while digital one & DAC, AMP are there just to let us fine granulate and adjust the sound signature which is important as not of the analogue sistems is perfect.
"Hhmm, I'm sure they have some experience as audiophiles"
The company that makes the card is called Audio Note. http://www.audionote.co.uk/ Been around a long time, with lots of high-end contracts for some of the biggest names in audio. It would be difficult to be *more* involved in the intricacies of audio.
Yes those that want high quality audio and hardware based features do buy audio cards. If you want basic or just above average audio then yes go out and buy a USB audio device. If onboard audio is good enough for some then sure go ahead use that since it came free on the board.
Quality audio cards have protections in place that shield out most everything and give you most times the best audio experience. Yes there are cheaper cards out there that suck both past and present and those you would want to stay away from. I am not sure about this card though being a USB to PCI-e type device it sounds a bit sketchy to me. But I am sure once it releases there will be reviews done on it and we will find out just how good/bad it is.
I have tried using onboard audio in the past and have always gone back to my hardware based card just because it sounds 100x better than anything I have tried with onboard audio and USB audio as well. I do however use a USB head phone setup but only when I am on Disc*rd all other times I am on my PCI-e audio hardware which is going into a audio/video head unit with large 5.1 speaker setup.
Ask a friend to do a blind test with you, and you will realize that you can't tell the difference between onboard audio and the most expensive card. Pretty much the only thing you can do to improve your audio experience is to buy better speakers.
Not to sure about that it is pretty easy to tell the difference between onboard and a full sound card. the first thing you will notice is background noise form other parts on the main board it might be very faint but it is there on most onboard sound setups. Besides that my friend might not notice the difference but I know I sure would.
I guess if you had just average speakers you might not notice because well the speakers are masking the lower quality audio form the onboard sound because they are not able to reproduce some of the audio spectrum the sound card is feeding to them. A high quality set of speakers will show the difference right away.
It all depends what you seek in your audio. If you simply want a card where you can customize how the card processes sound, a $50 Xonar allows you to swap OPAMPS, and would be adequate for most people who don't need a ultra-fancy DAC.
Where I'm at a loss is this card is essentially just a USB DAC. You can get a equivilent AK4493-based USB DAC (without swapable OPAMP) for $75 bucks. This will lack a decent 600ohm preAMP for headphones, though, so it really depends what your application is.
Meh, swapping opamps feels like a gimmick, to me. Could the noise introduced by having socketed op-amps possibly outweigh the potential improvement gained over whatever they could afford to ship in the device?
This is BS. It totally depends on the motherboard. On my Supermicro workstation board, the analog output has atrocious noise and cross-talk. It was clearly included merely for casual use with cheap desktop speakers.
Fortunately, it's also got an optical output.
Protip: make sure to mute *all* inputs. I mean: telephone, mic, CD (analog), line-in, etc. If there's noise when you aren't listening to anything and crank up the volume, this is the likely culprit.
Well there is difference, but you need good headphones to hear the difference and most motherboard based audio systems can not drive high quality headphones... If you use grap 50$ headphones... it does matter what sound system you use... even mp3 Sounds almost same as falck. But if you have 1000$ audiphile headphones you definitely hear the difference between mp3 and falck and good external dac vs onboard sound... well if the sound board can delivery sound to those headphones in anyway...
Motherboards and GPUs have digital audio outputs onboard (HDMI).
There is no difference in quality between an onboard S/PDIF or HDMI port and one on an audio card. Both their digital outputs are bit-perfect and they send the audio stream to my (far more expensive than an audio card) AVR with a nice quality DAC and nice quality amps in it.
The main difference is the multi-channel formats they can support. For whatever reason, the standardized formats supported by Toslink are quite limited. I think it's probably out of copy protection fears that once HDCP came onto the scene, all advancement in home theater moved to HDMI.
However, for stereo, toslink is fine. You can easily get 192 kHz 24-bit stereo, AFAIK.
I'm not quite sure what you mean by "hardware based features"... The only thing I can think of would be a 3D audio implementation, and those post-processing features are sort of hit-or-miss in my experience.
A sound card is by no means necessary for using an AV receiver, either. You can do it with onboard audio (like my desktop is) via optical, which has been a thing on motherboards for years at this point, or with a video card over HDMI (which is actually the better solution). In fact, depending on the interface you use to connect your sound card to your 5.1 surround sound system, a video card might work better.
It also sounds like you're used to "USB Audio" meaning "headset" instead of as an input for an actual DAC, which is fairly common nowadays. Every single DAC Schiit Audio makes (which range from $100 to $2400) features a USB audio input.
Optical (or coaxial, both use S/PDIF) can't support uncompressed surround sound. For a digital surround-sound interconnect, HDMI is absolutely superior to optical. *Technically* analog surround could match HDMI, but that's hardly supported anymore anyways.
I also don't use surround sound with an optical interconnect. The AV receiver I use over optical is acting as a *very* large headphone/stereo amplifier.
FWIW, Schiit actually used to sell DACs sans USB and positioned it as an add-on, and their designer still views it as an interior option to AES or even SPIDF even if some of the PR they put out around the latest USB implemention claim they solved most potential issues... I think they're still working on a further refined implementation beyond gen 5, tho that seems to work just as well as coax SPIDF for me.
I have an external one, and I love it. SoundBlaster G6. It has a quality headphone amp, decent sound settings, and most important of all has optical input with Dolby decoding, which works at the same time as my PC audio, so I can play a game on my PC or console while watching something on my TV or monitor and have both audio pumped into my headphones.
Agreed. I am sure this is some nice hardware, but after moving my audio processing to an external device, it is really hard to imagine going backwards. Do all the processing you want inside the box, but get it out via light pipe, and let your amp do what amps do best to your speakers or headphones. Not a lot of demand for internal cards any more... you would think they would know that?
IMO, the main use case for sound cards evaporated when people started doing audio processing on GPUs (and perhaps CPU cores with AVX). About 15-20 years ago, some sound card ASICs provided audio effects for games that would be impractical to do, otherwise. Now, if you build a sound card with enough processing power to rival what a GPU can handle, you'd pretty much have to put a GPU in it.
BTW, AMD briefly had an embedded Tensilica DSP to offload audio processing, in their GPUs. Now, they have better QoS features to enable using some subset of the main shader cores for it.
yep.. cause most onboard audio, doesnt have some of the features an add in soundcard has, unless you are going with a higher end motherboard. all but 1 of my comps i think.. all have soundcards, and on board sound is disabled....
Decent onboard audio is certainly possible, but I think it's often an area of cost-cutting for motherboards not aimed at a market where they can charge a premium for it or at least use it as a major selling point.
And inst it almost impossible to fully shield it if its on the PCI Express bus? I have never encountered an on board audio card that has been immune to having background noise that raised and fell with the load on the graphics card. Not since the original PCI audio card days at least.
In almost every way an external USB solution is better, worse is when they try to sell it as a gaming device that will be better than a USB solution, as if games still have native 3d audio processing on an audio cards processor.
Are DPC latency motherboard BIOS and driver issues still a thing? Because if so the last place I want my audio is hanging off a "PCIe to USB controller, making it an internal USB audio product."
It'd be nice if some of these PC audio solutions came with HDMI-audio-only output. For most PC setups that want to route digital surround sound solution (to a a/v receiver with passively powered speakers), the best fidelity is using digital audio passed through HDMI. Digital optical and digital coaxial have limited bandwidth and may not support more than 5 channels of audio (IIRC).
Can you route an HDMI port from a video card to a receiver? Yeah, you can, but you can't disable the "hidden" display associated with it (to prevent your mouse from wandering into that display associated with the a/v receiver) without also consequentially disabling the audio source being sent to the receiver. People literally make products to strip an HDMI signal to be audio only, when honestly that feature should be supported for any HDMI solution on a PC...
I get there's not much output space on the PCI-e slot cover for an HDMI out port, but if we're talking about a high end audio solution for PCs, how many really utilize the line in features? Sending analog audio in (phone/mp3 player?), to amplify it, to send amplified analog audio out? Why wouldn't you just transfer/play the music off the host PC with the sound card?
Same thing with Mic in port. The high quality interfaces for capturing microphone audio (we're assuming if someone has $$$ to buy a high quality soundcard, they likely also have $$$ to buy a high quality microphone) come in the form of USB audio interfaces that can take balanced XLR microphone input and sends that in as a digital signal over a USB bus. There's no real way to get an XLR input on a PCI-e device without making it a 2 slot monster. The other higher quality microphones on the market have an embedded USB interface such as the Blue line microphones (Yeti, Snowball, etc.) or the ATH-2020 USB, etc, so that a user doesn't need to buy a dedicated device for audio interfacing. All-in-all, 3.5mm output microphones on the market are generally not very good, and there's not a whole lot that a high-end soundcard can do to improve microphone quality if the source microphone has poor sensitivity or picks up too much noise.
IMO, i'd rather have high quality audio outputs available (such as hdmi-audio-only output for receivers + surround sound setups) than some audio inputs that wouldn't do a user much good.
Because of cost and it being easier to just do HDMI out via video card or your other video device, they're not going to bother with HDMI. They tried that at one point and there were issues with HDCP, so it was often problematic and didn't work well. I'm not sure if any of them ponied up to do Dolby TrueHD or DTS-HD decoding either (I think people thought they were when they were just doing pass through and then analog out of the standard DD/DTS), and certainly none tried Atmos and more recent formats. Would be nice to get a true object based digital audio processor, but
With regards to HDMI, I'd rather have input of it, so that you could capture it and maybe strip out the audio streams, but there's other devices that already do that for reasonable price so that's kinda limited in its appeal for an internal audio card.
I disagree on the mic input. I think it'd still matter. Absolutely the need for that is less due to the prevalence of USB ones, and yeah pro ones will use XLR, and there's plenty of pro digital interfaces to choose from. What they should do is make a USB-C based mic, that has its own ok ADC, but could plug into a USB-C port on a card that features higher end ADC, where they could leverage USB power but would just pass a balanced analog signal when paired with higher quality device. Maybe a wireless mic that could plug in via USB-C for higher quality. And the card have multiple USB-C inputs and work as sound board for say podcasters/streamers.
I think the only meaningful PC audio thing that most people would care about these days though, would be for someone to develop a very high end DSP designed for VR audio, where it would outdo PC software object based audio (but not be locked down like Creative did things that led to them squeezing them out). But it'd basically have to be paired with VR headsets (adding cost to them), or be cheap and open enough that it'd get consumer and developer support. But it could handle things like environment noise cancellation (to help immersion), give a bypass (for people talking to you). And it'd need some neat trick like being paired with some VR audio creation tool that lets you place sound channels where you want them in the 3D space, maybe with some like crazy Fantasia-esque symphonic conductor mode.
> I think the only meaningful PC audio thing that most people would care about these days though, would be for someone to develop a very high end DSP designed for VR audio, where it would outdo PC software object based audio
I'm the crazy one and use an Asus XG-C100C to move up to a 1024 audio channels in and out of my system.
As for this card, it appears to be fine high end consumer product but just doesn't seem like a professional card to me. There I'd expect balanced analog IO with similar high quality DACs and perhaps an actual DSP to tackle some of the audio processing. Sure, CPU's are fast enough for production processing but DSP's are useful and beneficial for lowend setups still. Beyond that would be some support for some network audio protocol like Dante or AVB. ASIO drivers are a must for pro usage too.
I'll wait to see what the drivers are like, if the card can work and perform with all features with base-Windows drivers, great! If it requires drivers from EVGA, then pass. Manufacturers always tend to bloat them like crazy and then abandon them after a couple years.
Plus for gaming, good luck trying to get every dev studio to QA this card to make sure it doesn't have issues with their game.
I'm speaking from Asus Xonar experience, thankfully the third-party drivers saved that hardware (for now anyway).
I use Behringer UCA202. Buy 10 at a time on ebay for $30 each. They have excellent test results. I have more expensive USB DAC's that except for pro balanced inputs are a comparative waste of money.
I'm trying to figure out why EVGA is deciding to get into sound cards right now. I do see a niche for sound cards (areas where space is at such a premium you can't even afford a tiny external DAC/Amp combo), but I don't anticipate the market growing at all.
In the internal sound card market, you have to challenge Creative, and I don't see what this offers that Creative's current flagship doesn't for $100 less. The only things that stand out to me are the RCA jacks for the line-output and DSD support, both of which are questionable inclusions to me.
To me, RCA line-level jacks implies that the analog output is going to be connected to some sort of amplifier. If so, why not use the optical out and an outboard DAC, too? As for DSD, it seems like the market for digital DSD files is actually than the market for SACD, which is tiny as it is.
The entire audiophile marketplace is filled with products that make little marketing sense. You could imagine a couple guys tearing down some piece of mass market hardware and talking about how they could do much better for just a little more $.
My guess is that the business case, if there is one, rests on bigger margins than the rest of their hardware. But maybe it's somebody's pet project and they don't really care if it at least breaks even.
The other thing I'm seeing is bundle deals with the rest of their hardware. Maybe they're planning on using this as the bait to help move their other hardware.
I'm with you on the outboard DAC. That's what I use. Maybe they will also offer an outboard variant of this.
Finally, remember that DSD could virtually blow up overnight, if someone like Spotify decides to offer it.
I don't think I can see DSD blowing up any time soon, especially not fron Spotify or another mass-market streaming service. Most DACs are designed around PCM, not DSD, and streaming service users (mostly) use their phones or a web browser to listen to music. They won't have a DAC that supports DSD, so the stream would need to be converted to PCM before it could be played. At that point, why not just stream 24/96 PCM files, which require *less bandwidth* uncompressed than 2.8MHz DSD?
(96 kHz * 24 bits per sample = 2.304 Mbps, 2.8 MHz * 1 bit per sample = 2.8 Mbps)
I wasn't predicting it *will* blow up, but just pointing out that in the age of apps and streaming, you don't have the same kind of inertia to overcome that physical discs and retail channels carried.
And when I say "blow up", I don't mean it would become massively popular. I just mean that a large catalog of content could become available in the format. So, you could imagine a streaming provider offering a premium "audiophile" service that provides their content in any one of multiple different audiophile formats.
i love how most people who are complaining about this soundcard are not the targeted customers...
The RCA output does not imply going to an amp. IT IS for an amp, and it's simple to understand. Flexibility, if you want to use desktop speakers instead of headphones, OR you want to use a different amp, solid state or tube, and still want to utilize the DAC. The AK4493 DAC is the latest and greatest. The highlights of this product are: AK4493 DAC, custom resistor and capacitors, and the analog volume control! Yes analog volume control.
While I do agree that an outboard DAC would be a better option, this product would cost 2x the price if it's configured that way. No, not 2x, I would say 3x. So in essence, pun intended if you get it, it's a "cheap" alternative for some folks, albeit a small market.
Anything in the "audiophile" market segment is not a big market, i'm sure EVGA has done their market research.
> analog volume control! Yes analog volume control.
This strikes me as so much audiophile silliness. With 32-bits per sample, there's no reason not to do digital volume control. Even 24-bit would be plenty, for that.
> While I do agree that an outboard DAC would be a better option, this product would cost 2x the price if it's configured that way. No, not 2x, I would say 3x.
And where does the added cost come from? Just the case and powersupply? Since it uses USB for connectivity, I don't imagine the control scheme would change any.
> Anything in the "audiophile" market segment is not a big market, i'm sure EVGA has done their market research.
Well, another pair of RCAs that you could gang together with external cables into balanced outputs would be nice. Not that I'm going to buy it either way, but if I would plug this thing directly into a power amp, I'd want to go balanced.
My current setup uses PC -(toslink)-> DAC -(balanced)-> Preamp/headphone-amp -(balanced)-> powered studio monitors.
My ideal setup would stay digital right up to the speakers. Then, the speakers can use a digital domain cross-over filter, EQ curve, and power amp that are optimally tuned for the speaker:
Asus has been making cards that compete with Creative's for years, maybe EVGA just thinks there's room for a third player... I don't remember paying more than $150 for my original Xonar STX but I know they keep updating them and the price points. I don't even use it for anything beyond Dolby Headphone DSP over coax SPIDF at this point, but it's not taking up any valuable room either since I finally went to a single GPU.
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nevcairiel - Tuesday, January 22, 2019 - link
Do people really still buy audio cards?If I wanted something high-end, I would go external, and avoid any electromagnetic interference inside the PC entirely, instead of hoping some vendors EMI shield actually works decently.
Ian Cutress - Tuesday, January 22, 2019 - link
The people behind the design are audiophiles that design custom solutions for the high end. If anyone knows what they're doing, it'll be themmlkj - Tuesday, January 22, 2019 - link
Hhmm, I'm sure they have some experience as audiophiles, but that doesn't stop them from taking the "bigger numbers are better game" too far.Let's take their dynamic range for example.
24-bits are enough to encode 140dB of dynamic range, and arguably even 16-bits with dithering reaches 120dB of dynamic range. And then their hardware only supports 123dB of dynamic range!
That 32bit audio support is just pretty misleading marketing then, if your hardware can't output a full 24bit by a wide margin.
But at least 32bit audio is not *worse* than 24 or 16. It's just not better.
The really dishonest trend is those 192kHz and 384kHz sample rates.
Counter-intuitively, higher sample rates is NOT something you want in digital audio =)
The short of it is that 44.1kHz and 48kHz are legitimately enough to perfectly (per the awesome Sampling theorem!) encode all sounds your hears can perceive, anything higher is ultrasonics that you can't hear, but that can and does add noise over the whole spectrum.
That audio is there and has to be processed when you chose a high sample rate, the problem is that audio hardware can't low-noise over the whole spectrum, not at any reasonable price.
Depending on how much your audio hardware prioritizes quality in those ultrasonic frequencies you can't hear (and that's unheard of, pun intended!) the noise might go from imperceptible to shockingly awful.
All in all the card might be good, the THD+N value looks totally acceptable for a PCIe card. But the truth is it won't be better quality than a cheap USB DAC.
If they know what they're doing like you say, then this card's trying to take advantage of audiophiles who don't understand all those numbers, and I can't recommend supporting this.
Angry internet comment over! =)
willis936 - Tuesday, January 22, 2019 - link
>Counter-intuitively, higher sample rates is NOT something you want in digital audioWell, this isn't really accurate. Higher sample rates are useful because they loosen the requirements of the analog filters in the DACs and ADCs, which are limiting factors. There is no audio information above 40 kSamp/s but if you want those dynamic range and SNR numbers then you need the higher sample rates.
mlkj - Tuesday, January 22, 2019 - link
I won't say there's no justification whatsoever for higher sample rates. Some margin is useful in analog filters (but again within reason, or any non-linearity near high-frequencies in your filters will further degrade the lower part of the spectrum).But at the end of the day I can't take those numbers at face value and tell people that this is a superior product.
I'm happy to claim that card's 384kHz/32bit output versus a similarly priced USB DAC would win in neither listening tests would nor objective noise metrics in the audible range.
nevcairiel - Tuesday, January 22, 2019 - link
The majority of DACs will internally oversample the input for filtering before analog conversion. Due to those attributes of most DACs, a higher input samplerate is not required.mode_13h - Tuesday, January 22, 2019 - link
And how good are those oversampling filters? I'll grant you than 384 kHz is overkill, but if the source material (e.g. mastering software, digital synths, etc.) can produce real signal at that rate, then oversampling can only be worse (i.e. by introducing artifacts).And for source material not at 384 kHz, your software can use whatever oversampling technique it wants, rather than being at the mercy of whatever the DAC does inside its little black box.
willis936 - Tuesday, January 22, 2019 - link
There's no reason to have source files above 48 kSamp/s. The extra 8 kSamp/s covers the non-linearities in the ADCs. Since all of the data you want to reproduce is 20 kHz and below there's nothing to be gained from higher frequencies. Upsampling is a very simple process and the minutiae of the interpolation filter doesn't really matter. It's all simple digital logic running at painfully low clock rates.mode_13h - Wednesday, January 23, 2019 - link
So, you're arguing against oversampling DACs? I think that argument was lost over 30 years ago.While not as common as 44.1 or 48 kHz, 96 and 192 kHz content has been with us since the mid 1990's. Since then, most recordings have probably been mastered in 24-bit/96 kHz or above, providing a massive catalog of potential content. These days, you don't need a new distribution format, like DVD Audio - you can simply add the format to new or existing streaming services.
And if you think the minutiae of the oversampling filters doesn't matter, then you're no audiophile. But, I think we've established that.
willis936 - Wednesday, January 23, 2019 - link
If you'd read the other comments you'll see that I'm not arguing against oversampling DACs. I'm arguing against oversampled content.Also throwing labels around isn't something I'm interested in, but since you've opened that up: no one should be proud to be an audiophile. Audiophile is a label that's been hijacked by a delusional group who throw objective measurements out the window because they don't want to put the effort in to understand what's going on. They're happy to be fooled and be taken for a ride. You're right. I am no audiophile.
The minutiae of the interpolation filters don't matter in that their audibility is incredibly low and as long as you're not picking something silly like NN or linear you won't see large performance differences. The differences will be there and you can go as far as picking neural net based algorithms, but you won't be able to hear the difference.
mode_13h - Wednesday, January 23, 2019 - link
Sure, there's a lot of stupidity in the audiophile community. I don't defend that. However, what you're missing is the ethos that one can only achieve the best audible performance, from an entire system, by optimizing each stage *beyond* the audible range. By definition, this can only be done through engineering, and measurement.And your assertion that the interpolation filter doesn't matter only applies to cases where the transition band is *completely* above the audible range (i.e. content that's already oversampled).
Finally, everyone here is forgetting that Shannon's Sampling Theorem only applies to *periodic* signals. Unless you sit around and listen to sine waves (or other repeating waveforms), audio data is not truly periodic. In other words, sampling theory is merely an approximation, for audio. Oversampling at (or close to) the source gives you additional margin for this, as well.
So, if you're trying to avoid over-engineering (and let's face it - the extra data we're talking about is cheap, by modern standards) and are satisfied with something that's merely decent, then good for you. Please just don't force your values on others.
CaedenV - Tuesday, January 22, 2019 - link
@willis936You are absolutely right. higher sample rates are not for the bits you can't hear, it is to make things as analog as possible before it hits the analog hardware. So, lets say (for sake of argument) that -1V is the bottom of a wave form, and 1V is the top of a wave form, and we have some 440 'waves per second' for a C note. Divide that up 44100 times and you end up with a whole series of voltage readings which can be used to reproduce that note. In digital, that is fine to reproduce something to the human ear, but to analog hardware (and especially for good hardware) this is a very long time between samples. The hardware can over-shoot, or under-shoot the next voltage reading which leads to weird jagged stair-stepping where it should be a nice even curve, which introduces weird artifacts in the music.
Up-sample that to 96-192Khz (beyond that is perhaps overkill), and now you have lots of voltages for the dac to read, and far less room for audible errors to be introduced by literally bored hardware that is just waiting to see what is next. Its not about creating tones that you cannot hear (but give dogs a more satisfying experience), it is about transitioning the stuff you can.
rpg1966 - Wednesday, January 23, 2019 - link
Caeden, that's not right. All you really need to think about is the #bits (which limits your DR), sample frequency (which limits your frequency response), and the filter at the end of the chain. When you start thinking in terms like "oh the stair step is too steppy", then you're not understanding how the filters work.mode_13h - Wednesday, January 23, 2019 - link
No, @CaedenV is exactly on the right track. In the 1980's and 1990's there seemed to be a race to reach ever-higher levels of oversampling, in DACs and CD players. I'm pretty sure I remember even reading about 256x oversampling, but 8x was not uncommon. I have a DAC from the late 90's with a Pacific Microsonics HDCD interface chip that used 8x oversampling.At a core level, the output voltage of PCM DACs is latched. This introduces "stair steps", like those having been noted. By increasing the sample rate, the noise introduced by this process is reduced in amplitude and pushed far outside the audible range, making it more easily addressed simple analog filters at the output.
https://en.wikipedia.org/wiki/Zero-order_hold
Kirby64 - Wednesday, January 23, 2019 - link
@mode_13h, you're wrong. Because analog filtering is on both sides (on the input for recording, on the output for playback), the 'stair step' you talk about doesn't ever actually exist. Yes, oversampling does technically allow you to use a crappier analog filter, but modern technology doesn't have this issue unless it's designed poorly. Once filtering is applied, the frequency will be accurately represented. This is just fundamentals of Nyquist-Shannon sampling theorem (https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shan...Watch this video: https://xiph.org/video/vid2.shtml - if you care about 'stair stepping', skip to chapter 3 specifically which addresses that.
The creator of that video goes into why sampling rate doesn't matter beyond say, 48 kHz.
mode_13h - Wednesday, January 23, 2019 - link
*sigh* this is kid stuff. Covered in the first chapter of any signal processing text.Yes, you *do* need to band-limit the signal, before sampling, on the upstream side. No argument there. Failure to do that will result in aliasing, which is impossible to remove, unless you can make some assumptions about the signal (in which case you're also not actually using the full bandwidth of the channel).
Second, he's glossing over what the device is doing to groom the output. His test would be relevant to my point if he had a PCM DAC chip on a breadboard - not testing an end product.
Finally, see my above point about sampling theorem vs. audio. He's using sine waves, since they're easy to understand and work with. How one can actually quantify the performance of a system is by passing through a full-bandwidth signal and subtract it off (or divide it by) the original. That's how you'd be able to observe any corners being cut along the way.
Here's something a bit more worthwhile:
http://bigwww.epfl.ch/publications/unser0001.pdf
slides: https://basesandframes.files.wordpress.com/2016/05...
mode_13h - Wednesday, January 23, 2019 - link
This explains the context of the slides. They were prepared to describe the paper, but not by its author.https://basesandframes.wordpress.com/2016/05/12/50...
Kirby64 - Thursday, January 24, 2019 - link
Are we not talking about complete systems here? I'm sure if you use some raw DAC with no filtering then weird stuff can happen. No question there. Then you aren't properly band-limiting a system.Your above post talking about sampling theorem vs. audio is incorrect. Applicability is to periodic OR non-periodic signals. I can cram as much as I want within the bandwidth of a device and it should be able to be accurately reproduced. If I have a single step waveform sampled the only way to actually represent that is with ripples and a decay that would be in accordance with the sampling rate I have available. There's only one pattern of signals that would satisfy that waveform. Yes you have to worry about intermod distortion and other non-linear effects when you start throwing more and more signal content in there, nothing is perfect, but these issues are relatively minor.
What are you actually claiming is the benefit of oversampling DAC in relation to the output voltage? Closer accuracy to the signal because there's more points as it 'approximates' it? With a filter that doesn't matter.
I'm not sure how higher bit depth or higher sampling frequency fix any of the problems with 'corners being cut'. I'm sure I can design a crap 384kHz 32-bit DAC and an amazing 16-bit 48kHz DAC; the marketing numbers don't really have anything to do with how good it actually performs.
mode_13h - Thursday, January 24, 2019 - link
I was specifically talking about what comes out of a PCM DAC, prior to analog filtering, which you incorrectly said didn't resemble a stair step.https://en.wikipedia.org/wiki/Zero-order_hold
Regarding sampling, you're contradicting yourself. Shannon says you can't have aliasing if you stay below the Nyquist limit, but that's only for periodic waveforms. Now, you're trying to add a bunch of caveats and tell me that I can't *really* use the entire channel bandwidth? Just read the paper I linked and bring yourself up to date, circa 20 years ago.
The author also made presentation on the same subject material, 10 years after that:
http://citeseerx.ist.psu.edu/viewdoc/download?doi=...
And what I claim is the obvious - of course the quality of the interpolation filter matters! And no, your analog output filter is not magic. It can't make a non-oversampled signal/DAC perform as well, and it doesn't make the quality of the oversampling filter irrelevant.
I suppose you've never heard of noise shaping, either.
> I'm sure I can design a crap 384kHz 32-bit DAC and an amazing 16-bit 48kHz DAC; the marketing numbers don't really have anything to do with how good it actually performs.
It's cheaper and easier to get good performance with an oversampled signal. That was the whole point of delta/sigma DACs, but it also holds true of PCM DACs. You know enough to have opinions, but you're no DSP engineer.
As for bit-depth, it depends on the original signal. 16-bit doesn't quite match the dynamic range of human hearing. For home listening purposes, what's interesting with 24-bit (or more) is you can actually cut out the analog preamp and run your DAC straight into your power amp, doing your volume control in the digital domain.
mode_13h - Thursday, January 24, 2019 - link
I forgot to add that bit depth also affects phase accuracy. Of course, if your input is only 16-bit, then oversampling that obviously won't improve accuracy that's already been lost (though it can mitigate the impact of downstream filtering).This brings up an interesting point, however. Because the high frequencies in music tend to be fairly low amplitude, their effective bit depth is lower. To address this CDs have a feature called pre-emphasis, which is essentially an EQ curve that boost high frequencies during mastering, and a flag telling the CD player to attenuate them after the DAC (although CD players with oversampling and/or higher native bit-depth DACs can do it in the digital domain).
https://en.wikipedia.org/wiki/Emphasis_(telecommun...
I have some CDs with pre-emphasis and they sound great. Excellent highs and reverb. I wish it had caught on, more.
Kirby64 - Friday, January 25, 2019 - link
I think we may actually be more in agreement than not in terms of actual hardware info - we're just approaching this from different lenses. I care about full systems here and what the marketing wank numbers actually mean to my audio quality. What the chips do internally is often times hidden from users. Using a delta-sig (that internally does over-sampling) to achieve a better bit depth as a result is fine by me. It's just a means to an end. My point being that the label of 24 bit or 384kHz sampling (or even 192kHz sampling) is definitively not a label that must mean 'good audio quality'. There's a ton of other factors at play. Usually the fact that my audio doesn't have a bunch of interference from nearby noise sources plays a bigger role than all this other nonsense.That being said, any single audio DAC chip won't look like a stair step on the actual output due to inherent limitations of slew rate/high freq attenuation internally. Also, any audio chip you buy worth it's salt is going to have filtering already built into it. Most seem to quote at least 100dB of attenuation at ~0.45*Fs.
Also, not sure about your point on sampling? Where did I say you can't use the entire channel bandwidth? My only point is that non-periodic signals (to be 'truly' represented) effectively need infinite bandwidth to represent. Sure, after only a short period of time, the need for the extra bandwidth is lost in noise, but if we live in the perfect world (where DACs have nice sharp edges on their stair step output) then you need infinite bandwidth.
Also, it's definitely not cheaper to get good performance out of a higher bit depth DAC. Silicon costs money, full stop. If you can design a DAC that performs just fine with 16-bits and lower sampling rate, that is going to cost less. Unless you mean engineering investment to actually construct that device... then maybe there's a tradeoff.
Lastly, I think we'll have to agree to disagree on 16-bit vs. 24-bit usefulness for listening though. Same can be said of emphasis additions. There's plenty of tests that show people can't tell the difference between 16-bit and 24-bit well constructed audio.
mode_13h - Friday, January 25, 2019 - link
I agree that sample rate and bit depth cannot be used as a simple proxy for the audio quality of a device.Second, I figured most audio DACs have built-in analog filtering, for probably a long time now. Probably also oversampling, as well. That doesn't make my point invalid, as I'm concerned with the burden sample rate places on the design and performance of the analog output filter - whether or not it's integrated.
My point about periodic vs. non-periodic signals is that adding more bandwidth should allow more headroom to handle the transients other aspects where audio signals behave differently than truly periodic waveforms. I think 96 kHz should provide plenty of margin.
I never said that a DAC with a larger word length would be cheaper. Again, I'm concerned with sample rate. And to that end, the reason higher sample rates and oversampling enable better output quality per $ is due to simplifying the demands on the analog output filter. With something like 96 kHz, you can use an antialias filter with a wide transition band, simplifying the task of minimizing resonance and maintaining good phase linearity.
Most of my audio library is 44.1 @ 16-bit, and I enjoy it quite a bit. I think where you're likely to hear the limitations of that format is in the corner cases. Also, we don't know how much recording engineers are constrained by it, in production (though, only a concern for less "pop" recordings).
BTW, one of the infuriating things about comparing CDs to vinyl or DVD audio is that the mix is usually different. So, you're not really comparing just the media or format. But I never bought into the vinyl hype, and its copy protection kept me away from DVD audio.
darkswordsman17 - Tuesday, January 22, 2019 - link
They're just citing the figures of the DAC/ADC, and pretty much no actual products ever meet those specs, so don't read too much into that stuff. Even pro stuff doesn't typically come close to the rated specs of the DACs (some can, but most are probably a good 10-20dB below the rated spec of the DAC). If its 100dB or above its decent. Oh and I'm talking about the final signal post analog section (as its not difficult to get the rated spec out of just the digital realm).Yes 32 bits is rather pointless and largely marketing fluff (it'd be useful in audio production and for archiving or something, but is largely pointless for consumers), just like the stupid high sample rates (last I saw they were up to 768kHz, and are at 22.xxx on DSD stuff). But that's just how audio is these days sadly. You should look into DSD vs PCM, and even Delta Sigma vs multi-bit (R2R for instance) and similar discussions raging in the audiophile world these days. They're desperate to keep pushing people to upgrade. There's one company that touted its "medical grade" multi-bit DAC chips that they then admitted (on forums) test much worse than even mid-range PCM DACs for audio (because those multi-bit DACs are not being used for audio in the medical field, and the medical equipment using it are expensive for reasons other than the "it meets the specs" DAC chips used), but still try to claim that its world beating audio quality. This company has a cult following. They make some decent stuff, but despite them claiming to be beyond audiophile nonsense, some of their products often don't meet basic electronics (not properly grounded, they had one that was unloading DC feedback when turned on, which was visibly distorting headphone speaker diaphragms; to be fair they did fix those issues in production on some of it, but somehow new products still come out with some of the same issues from time to time - and its not because "bad run" during production, its because they'd overlook stuff like that during the engineering phase).
I won't say it wouldn't be better (there are other factors, and yes, some like shielding from noise that will be to the USB DAC favor), as there's other issues. I've used a variety of external audio gear and keep going back to my X-Fi Titanium HD (which this card is pretty similar to, high quality ADC and DAC). It has some flaws to be sure (can definitely get some noise from other components from time to time although that's been rare, and sometimes due to Windows Updates it'll reset the software settings - which is discernible via listening - I turn off all the processing of the card and put it in Audio Production mode so that it just passes bits to the DAC and then analog out), but still sounds good to me. I've tried a variety of USB DACs and keep going back to just the Titanium HD. This included some popular with audiophile DSD and multi-bit DACs, and neither of them sounded right to me, and something audiophiles seem to want to refuse to accept is that both types of devices (DSD and "multi-bit") tend to do a lot of shaping and filtering of audio signal and that is what they're hearing (so its not "true to the actual audio signal" like some of them claim). And that they could accomplish the same with PCM (which PCM does some things of its own, and some audiophiles know that and just claim that they prefer the filtering of one over the other, or often the specific implementation of some piece of expensive audiophile DAC they bought), but they'll frown about EQ and other "processing" calling it fake and "digital sounding" and other fairly nonsensical phrasing. And some programs have been working to add some of the distortion that some people find appealing that is cited for vinyl and/or tube based "analog sound" that some try to claim is why they sound better.
mlkj - Tuesday, January 22, 2019 - link
Yep, I think that's fair enough.mode_13h - Tuesday, January 22, 2019 - link
Ironically, delta/sigma DACs originated as a lower-cost alternative to PCM. Then, I guess DSD is a consequence of some folks getting carried away with the idea.The funny thing is that I'm pretty sure even DSD recordings are mastered (if not also recorded) using PCM-based software. So, you're doing at least one inexact conversion, which might not have audible artifacts but I'm sure it has some.
mode_13h - Tuesday, January 22, 2019 - link
> The really dishonest trend is those 192kHz and 384kHz sample rates. Counter-intuitively, higher sample rates is NOT something you want in digital audio =)Where are you getting that? Sounds a lot like an invented fact. Why do you even care about low noise, in ultrasonic frequency bands? You won't hear it, by definition, so it's no matter if a little is there.
As said by others, a higher filter rate improves the effective performance of low-pass antialiasing filters. You can put the entire transition band above the audible range, along with any resonance and leakage. Filter design is a game of compromises. Oversampling eases the burden, substantially.
I'll grant you that 384 kHz at the input is overkill, but DACs have long oversampled to such frequencies and beyond.
https://en.wikipedia.org/wiki/Oversampling
> it won't be better quality than a cheap USB DAC
I doubt that. It sounds like their power filtering is substantially better, if not also their output stage.
zodiacfml - Tuesday, January 22, 2019 - link
YesZolaIII - Wednesday, January 23, 2019 - link
Let's just say how 96000Hz at 24 bits is just perfect enough. You need a higher range than you can here so that you can parametric adjust the high range that you can hear but that it doesn't interfere with lower high range & mids. Still real magic lies in getting the range from 128 to 600 Hz (vocals) right & to push the 12~19 Hz range (sub bass) high enough to be auditable. SNR is single most important metric along with actual dynamics (dB) range that makes what we call sound stage and separation. Still analog end listening device is the one most important while digital one & DAC, AMP are there just to let us fine granulate and adjust the sound signature which is important as not of the analogue sistems is perfect.Best regards.
Ian Cutress - Thursday, January 24, 2019 - link
"Hhmm, I'm sure they have some experience as audiophiles"The company that makes the card is called Audio Note.
http://www.audionote.co.uk/
Been around a long time, with lots of high-end contracts for some of the biggest names in audio. It would be difficult to be *more* involved in the intricacies of audio.
rocky12345 - Tuesday, January 22, 2019 - link
"Do people really still buy audio cards?"Yes those that want high quality audio and hardware based features do buy audio cards. If you want basic or just above average audio then yes go out and buy a USB audio device. If onboard audio is good enough for some then sure go ahead use that since it came free on the board.
Quality audio cards have protections in place that shield out most everything and give you most times the best audio experience. Yes there are cheaper cards out there that suck both past and present and those you would want to stay away from. I am not sure about this card though being a USB to PCI-e type device it sounds a bit sketchy to me. But I am sure once it releases there will be reviews done on it and we will find out just how good/bad it is.
I have tried using onboard audio in the past and have always gone back to my hardware based card just because it sounds 100x better than anything I have tried with onboard audio and USB audio as well. I do however use a USB head phone setup but only when I am on Disc*rd all other times I am on my PCI-e audio hardware which is going into a audio/video head unit with large 5.1 speaker setup.
p1esk - Tuesday, January 22, 2019 - link
Ask a friend to do a blind test with you, and you will realize that you can't tell the difference between onboard audio and the most expensive card. Pretty much the only thing you can do to improve your audio experience is to buy better speakers.rocky12345 - Tuesday, January 22, 2019 - link
Not to sure about that it is pretty easy to tell the difference between onboard and a full sound card. the first thing you will notice is background noise form other parts on the main board it might be very faint but it is there on most onboard sound setups. Besides that my friend might not notice the difference but I know I sure would.I guess if you had just average speakers you might not notice because well the speakers are masking the lower quality audio form the onboard sound because they are not able to reproduce some of the audio spectrum the sound card is feeding to them. A high quality set of speakers will show the difference right away.
Samus - Tuesday, January 22, 2019 - link
It all depends what you seek in your audio. If you simply want a card where you can customize how the card processes sound, a $50 Xonar allows you to swap OPAMPS, and would be adequate for most people who don't need a ultra-fancy DAC.Where I'm at a loss is this card is essentially just a USB DAC. You can get a equivilent AK4493-based USB DAC (without swapable OPAMP) for $75 bucks. This will lack a decent 600ohm preAMP for headphones, though, so it really depends what your application is.
mode_13h - Wednesday, January 23, 2019 - link
Meh, swapping opamps feels like a gimmick, to me. Could the noise introduced by having socketed op-amps possibly outweigh the potential improvement gained over whatever they could afford to ship in the device?Dubbo07 - Wednesday, January 23, 2019 - link
It is as gimmicky as switching out a 8700k to a 9700k cpu.mode_13h - Wednesday, January 23, 2019 - link
LOL, but I don't buy it. I'd want to see the performance of the socketed version quantified and compared to soldered, first.mode_13h - Tuesday, January 22, 2019 - link
This is BS. It totally depends on the motherboard. On my Supermicro workstation board, the analog output has atrocious noise and cross-talk. It was clearly included merely for casual use with cheap desktop speakers.Fortunately, it's also got an optical output.
Protip: make sure to mute *all* inputs. I mean: telephone, mic, CD (analog), line-in, etc. If there's noise when you aren't listening to anything and crank up the volume, this is the likely culprit.
haukionkannel - Wednesday, January 23, 2019 - link
Well there is difference, but you need good headphones to hear the difference and most motherboard based audio systems can not drive high quality headphones...If you use grap 50$ headphones... it does matter what sound system you use... even mp3 Sounds almost same as falck. But if you have 1000$ audiphile headphones you definitely hear the difference between mp3 and falck and good external dac vs onboard sound... well if the sound board can delivery sound to those headphones in anyway...
nevcairiel - Tuesday, January 22, 2019 - link
You obviously have never used a professional-grade USB DAC, you seem to think those are limited to cheap USB headphones.mode_13h - Wednesday, January 23, 2019 - link
Lol, you've got to use AES/EBU or glass ST optical, or else it's not "professional".:)
SirMaster - Tuesday, January 22, 2019 - link
Motherboards and GPUs have digital audio outputs onboard (HDMI).There is no difference in quality between an onboard S/PDIF or HDMI port and one on an audio card. Both their digital outputs are bit-perfect and they send the audio stream to my (far more expensive than an audio card) AVR with a nice quality DAC and nice quality amps in it.
mode_13h - Wednesday, January 23, 2019 - link
The main difference is the multi-channel formats they can support. For whatever reason, the standardized formats supported by Toslink are quite limited. I think it's probably out of copy protection fears that once HDCP came onto the scene, all advancement in home theater moved to HDMI.However, for stereo, toslink is fine. You can easily get 192 kHz 24-bit stereo, AFAIK.
Inteli - Tuesday, January 22, 2019 - link
I'm not quite sure what you mean by "hardware based features"... The only thing I can think of would be a 3D audio implementation, and those post-processing features are sort of hit-or-miss in my experience.A sound card is by no means necessary for using an AV receiver, either. You can do it with onboard audio (like my desktop is) via optical, which has been a thing on motherboards for years at this point, or with a video card over HDMI (which is actually the better solution). In fact, depending on the interface you use to connect your sound card to your 5.1 surround sound system, a video card might work better.
It also sounds like you're used to "USB Audio" meaning "headset" instead of as an input for an actual DAC, which is fairly common nowadays. Every single DAC Schiit Audio makes (which range from $100 to $2400) features a USB audio input.
mode_13h - Tuesday, January 22, 2019 - link
AFAIK, multi-channel optical is limited to 48 kHz @ 5.1.HDMI can support more channels at higher bit-rates and sample-rates. Also, more/newer codecs. The main downside is thicker, shorter cables.
Inteli - Wednesday, January 23, 2019 - link
Optical (or coaxial, both use S/PDIF) can't support uncompressed surround sound. For a digital surround-sound interconnect, HDMI is absolutely superior to optical. *Technically* analog surround could match HDMI, but that's hardly supported anymore anyways.I also don't use surround sound with an optical interconnect. The AV receiver I use over optical is acting as a *very* large headphone/stereo amplifier.
Impulses - Tuesday, March 5, 2019 - link
FWIW, Schiit actually used to sell DACs sans USB and positioned it as an add-on, and their designer still views it as an interior option to AES or even SPIDF even if some of the PR they put out around the latest USB implemention claim they solved most potential issues... I think they're still working on a further refined implementation beyond gen 5, tho that seems to work just as well as coax SPIDF for me.mode_13h - Tuesday, January 22, 2019 - link
Since I could get bit-perfect optical out, I've gone with outboard DACs.I use separates, but here's one I always thought was cute:
http://www.parasound.com/zdac-v2.php
Sttm - Tuesday, January 22, 2019 - link
I have an external one, and I love it. SoundBlaster G6. It has a quality headphone amp, decent sound settings, and most important of all has optical input with Dolby decoding, which works at the same time as my PC audio, so I can play a game on my PC or console while watching something on my TV or monitor and have both audio pumped into my headphones.CaedenV - Tuesday, January 22, 2019 - link
Agreed. I am sure this is some nice hardware, but after moving my audio processing to an external device, it is really hard to imagine going backwards. Do all the processing you want inside the box, but get it out via light pipe, and let your amp do what amps do best to your speakers or headphones. Not a lot of demand for internal cards any more... you would think they would know that?mode_13h - Tuesday, January 22, 2019 - link
IMO, the main use case for sound cards evaporated when people started doing audio processing on GPUs (and perhaps CPU cores with AVX). About 15-20 years ago, some sound card ASICs provided audio effects for games that would be impractical to do, otherwise. Now, if you build a sound card with enough processing power to rival what a GPU can handle, you'd pretty much have to put a GPU in it.BTW, AMD briefly had an embedded Tensilica DSP to offload audio processing, in their GPUs. Now, they have better QoS features to enable using some subset of the main shader cores for it.
https://en.wikipedia.org/wiki/AMD_TrueAudio
Qasar - Tuesday, January 22, 2019 - link
" Do people really still buy audio cards? "yep.. cause most onboard audio, doesnt have some of the features an add in soundcard has, unless you are going with a higher end motherboard. all but 1 of my comps i think.. all have soundcards, and on board sound is disabled....
mode_13h - Wednesday, January 23, 2019 - link
Decent onboard audio is certainly possible, but I think it's often an area of cost-cutting for motherboards not aimed at a market where they can charge a premium for it or at least use it as a major selling point.Qasar - Wednesday, January 23, 2019 - link
its.,. but as i mentioned... " higher end motherboard " :-)Krause - Friday, January 25, 2019 - link
And inst it almost impossible to fully shield it if its on the PCI Express bus? I have never encountered an on board audio card that has been immune to having background noise that raised and fell with the load on the graphics card. Not since the original PCI audio card days at least.In almost every way an external USB solution is better, worse is when they try to sell it as a gaming device that will be better than a USB solution, as if games still have native 3d audio processing on an audio cards processor.
Gunbuster - Tuesday, January 22, 2019 - link
Are DPC latency motherboard BIOS and driver issues still a thing? Because if so the last place I want my audio is hanging off a "PCIe to USB controller, making it an internal USB audio product."JoeyJoJo123 - Tuesday, January 22, 2019 - link
It'd be nice if some of these PC audio solutions came with HDMI-audio-only output. For most PC setups that want to route digital surround sound solution (to a a/v receiver with passively powered speakers), the best fidelity is using digital audio passed through HDMI. Digital optical and digital coaxial have limited bandwidth and may not support more than 5 channels of audio (IIRC).Can you route an HDMI port from a video card to a receiver? Yeah, you can, but you can't disable the "hidden" display associated with it (to prevent your mouse from wandering into that display associated with the a/v receiver) without also consequentially disabling the audio source being sent to the receiver. People literally make products to strip an HDMI signal to be audio only, when honestly that feature should be supported for any HDMI solution on a PC...
https://www.monoprice.com/Product?c_id=101&cp_...
I get there's not much output space on the PCI-e slot cover for an HDMI out port, but if we're talking about a high end audio solution for PCs, how many really utilize the line in features? Sending analog audio in (phone/mp3 player?), to amplify it, to send amplified analog audio out? Why wouldn't you just transfer/play the music off the host PC with the sound card?
Same thing with Mic in port. The high quality interfaces for capturing microphone audio (we're assuming if someone has $$$ to buy a high quality soundcard, they likely also have $$$ to buy a high quality microphone) come in the form of USB audio interfaces that can take balanced XLR microphone input and sends that in as a digital signal over a USB bus. There's no real way to get an XLR input on a PCI-e device without making it a 2 slot monster. The other higher quality microphones on the market have an embedded USB interface such as the Blue line microphones (Yeti, Snowball, etc.) or the ATH-2020 USB, etc, so that a user doesn't need to buy a dedicated device for audio interfacing. All-in-all, 3.5mm output microphones on the market are generally not very good, and there's not a whole lot that a high-end soundcard can do to improve microphone quality if the source microphone has poor sensitivity or picks up too much noise.
IMO, i'd rather have high quality audio outputs available (such as hdmi-audio-only output for receivers + surround sound setups) than some audio inputs that wouldn't do a user much good.
darkswordsman17 - Tuesday, January 22, 2019 - link
Because of cost and it being easier to just do HDMI out via video card or your other video device, they're not going to bother with HDMI. They tried that at one point and there were issues with HDCP, so it was often problematic and didn't work well. I'm not sure if any of them ponied up to do Dolby TrueHD or DTS-HD decoding either (I think people thought they were when they were just doing pass through and then analog out of the standard DD/DTS), and certainly none tried Atmos and more recent formats. Would be nice to get a true object based digital audio processor, butWith regards to HDMI, I'd rather have input of it, so that you could capture it and maybe strip out the audio streams, but there's other devices that already do that for reasonable price so that's kinda limited in its appeal for an internal audio card.
I disagree on the mic input. I think it'd still matter. Absolutely the need for that is less due to the prevalence of USB ones, and yeah pro ones will use XLR, and there's plenty of pro digital interfaces to choose from. What they should do is make a USB-C based mic, that has its own ok ADC, but could plug into a USB-C port on a card that features higher end ADC, where they could leverage USB power but would just pass a balanced analog signal when paired with higher quality device. Maybe a wireless mic that could plug in via USB-C for higher quality. And the card have multiple USB-C inputs and work as sound board for say podcasters/streamers.
I think the only meaningful PC audio thing that most people would care about these days though, would be for someone to develop a very high end DSP designed for VR audio, where it would outdo PC software object based audio (but not be locked down like Creative did things that led to them squeezing them out). But it'd basically have to be paired with VR headsets (adding cost to them), or be cheap and open enough that it'd get consumer and developer support. But it could handle things like environment noise cancellation (to help immersion), give a bypass (for people talking to you). And it'd need some neat trick like being paired with some VR audio creation tool that lets you place sound channels where you want them in the 3D space, maybe with some like crazy Fantasia-esque symphonic conductor mode.
mode_13h - Tuesday, January 22, 2019 - link
> I think the only meaningful PC audio thing that most people would care about these days though, would be for someone to develop a very high end DSP designed for VR audio, where it would outdo PC software object based audioMaybe something like this?
https://gpuopen.com/gaming-product/true-audio-next...
https://developer.nvidia.com/vrworks/vrworks-audio
However, offloading noise cancellation to the GPU would almost certainly incur too much latency and jitter.
willis936 - Tuesday, January 22, 2019 - link
These things seem more useful as instruments than for actual audio.Kevin G - Tuesday, January 22, 2019 - link
I'm the crazy one and use an Asus XG-C100C to move up to a 1024 audio channels in and out of my system.As for this card, it appears to be fine high end consumer product but just doesn't seem like a professional card to me. There I'd expect balanced analog IO with similar high quality DACs and perhaps an actual DSP to tackle some of the audio processing. Sure, CPU's are fast enough for production processing but DSP's are useful and beneficial for lowend setups still. Beyond that would be some support for some network audio protocol like Dante or AVB. ASIO drivers are a must for pro usage too.
BinaryTB - Tuesday, January 22, 2019 - link
I'll wait to see what the drivers are like, if the card can work and perform with all features with base-Windows drivers, great! If it requires drivers from EVGA, then pass. Manufacturers always tend to bloat them like crazy and then abandon them after a couple years.Plus for gaming, good luck trying to get every dev studio to QA this card to make sure it doesn't have issues with their game.
I'm speaking from Asus Xonar experience, thankfully the third-party drivers saved that hardware (for now anyway).
coit - Tuesday, January 22, 2019 - link
I use Behringer UCA202. Buy 10 at a time on ebay for $30 each. They have excellent test results.I have more expensive USB DAC's that except for pro balanced inputs are a comparative waste of money.
coit - Tuesday, January 22, 2019 - link
Correction: balanced outputsrpg1966 - Wednesday, January 23, 2019 - link
"The Nu Audio card is equipped with two RCA line outs for left and right speakers that can output 384 KHz at 32-bit audio,"Line-outs output analogue signals, not xxKHz/yy-bits /pedant
Inteli - Wednesday, January 23, 2019 - link
They're clearly talking about the DAC behind the line-level outputs, which is important information.Inteli - Wednesday, January 23, 2019 - link
I'm trying to figure out why EVGA is deciding to get into sound cards right now. I do see a niche for sound cards (areas where space is at such a premium you can't even afford a tiny external DAC/Amp combo), but I don't anticipate the market growing at all.In the internal sound card market, you have to challenge Creative, and I don't see what this offers that Creative's current flagship doesn't for $100 less. The only things that stand out to me are the RCA jacks for the line-output and DSD support, both of which are questionable inclusions to me.
To me, RCA line-level jacks implies that the analog output is going to be connected to some sort of amplifier. If so, why not use the optical out and an outboard DAC, too? As for DSD, it seems like the market for digital DSD files is actually than the market for SACD, which is tiny as it is.
mode_13h - Wednesday, January 23, 2019 - link
The entire audiophile marketplace is filled with products that make little marketing sense. You could imagine a couple guys tearing down some piece of mass market hardware and talking about how they could do much better for just a little more $.My guess is that the business case, if there is one, rests on bigger margins than the rest of their hardware. But maybe it's somebody's pet project and they don't really care if it at least breaks even.
The other thing I'm seeing is bundle deals with the rest of their hardware. Maybe they're planning on using this as the bait to help move their other hardware.
I'm with you on the outboard DAC. That's what I use. Maybe they will also offer an outboard variant of this.
Finally, remember that DSD could virtually blow up overnight, if someone like Spotify decides to offer it.
Inteli - Wednesday, January 23, 2019 - link
I don't think I can see DSD blowing up any time soon, especially not fron Spotify or another mass-market streaming service. Most DACs are designed around PCM, not DSD, and streaming service users (mostly) use their phones or a web browser to listen to music. They won't have a DAC that supports DSD, so the stream would need to be converted to PCM before it could be played. At that point, why not just stream 24/96 PCM files, which require *less bandwidth* uncompressed than 2.8MHz DSD?(96 kHz * 24 bits per sample = 2.304 Mbps, 2.8 MHz * 1 bit per sample = 2.8 Mbps)
mode_13h - Thursday, January 24, 2019 - link
I wasn't predicting it *will* blow up, but just pointing out that in the age of apps and streaming, you don't have the same kind of inertia to overcome that physical discs and retail channels carried.And when I say "blow up", I don't mean it would become massively popular. I just mean that a large catalog of content could become available in the format. So, you could imagine a streaming provider offering a premium "audiophile" service that provides their content in any one of multiple different audiophile formats.
Dubbo07 - Saturday, January 26, 2019 - link
i love how most people who are complaining about this soundcard are not the targeted customers...The RCA output does not imply going to an amp. IT IS for an amp, and it's simple to understand.
Flexibility, if you want to use desktop speakers instead of headphones, OR you want to use a different amp, solid state or tube, and still want to utilize the DAC. The AK4493 DAC is the latest and greatest. The highlights of this product are: AK4493 DAC, custom resistor and capacitors, and the analog volume control! Yes analog volume control.
While I do agree that an outboard DAC would be a better option, this product would cost 2x the price if it's configured that way. No, not 2x, I would say 3x. So in essence, pun intended if you get it, it's a "cheap" alternative for some folks, albeit a small market.
Anything in the "audiophile" market segment is not a big market, i'm sure EVGA has done their market research.
mode_13h - Saturday, January 26, 2019 - link
> analog volume control! Yes analog volume control.This strikes me as so much audiophile silliness. With 32-bits per sample, there's no reason not to do digital volume control. Even 24-bit would be plenty, for that.
> While I do agree that an outboard DAC would be a better option, this product would cost 2x the price if it's configured that way. No, not 2x, I would say 3x.
And where does the added cost come from? Just the case and powersupply? Since it uses USB for connectivity, I don't imagine the control scheme would change any.
> Anything in the "audiophile" market segment is not a big market, i'm sure EVGA has done their market research.
Well, another pair of RCAs that you could gang together with external cables into balanced outputs would be nice. Not that I'm going to buy it either way, but if I would plug this thing directly into a power amp, I'd want to go balanced.
My current setup uses PC -(toslink)-> DAC -(balanced)-> Preamp/headphone-amp -(balanced)-> powered studio monitors.
My ideal setup would stay digital right up to the speakers. Then, the speakers can use a digital domain cross-over filter, EQ curve, and power amp that are optimally tuned for the speaker:
https://www.genelec.com/key-technologies/smart-act...
Impulses - Tuesday, March 5, 2019 - link
Asus has been making cards that compete with Creative's for years, maybe EVGA just thinks there's room for a third player... I don't remember paying more than $150 for my original Xonar STX but I know they keep updating them and the price points. I don't even use it for anything beyond Dolby Headphone DSP over coax SPIDF at this point, but it's not taking up any valuable room either since I finally went to a single GPU.