$50 Sound Cards Impress Versus Integrated Audio

crookedvulture writes "Most PCs have audio integrated right on the motherboard. There's much to be gained from upgrading to a discrete sound card, though. This look at a couple of sub-$50 sound cards from Asus explores what can be found at the budget end of the spectrum. In blind listening tests, both cards produced better sound than an integrated solution. They also offered superior signal quality, but neither had an impact on gaming performance. The days of hardware-accelerated game audio seem to be behind us, with developers handling positional audio processing in software."

Comments

Distance from the power supply

[itamblyn]

Most of the improvement is likely due to increased distance between the amplification circuits and the noisy AC/DC power supply.

Re:Distance from the power supply

[O('_')O_Bush]

That, and larger circuits/ higher power means noise impacts less.

Re:Distance from the power supply

[ChrisMP1]

A decent ATX power supply really isn't bad. Distance from the noisy digital circuitry on the motherboard is important, though.

Re:Distance from the power supply

[TheGratefulNet]

sometimes the circuit (board) does this work.

case in point, ESI juli@ (hate the at-sign. how stupid!)

this card is lab grade (test gear quality) in its a/d and d/a. some people use it for RMAA audio gear testing. not kidding! this is a low noise floor that you won't normally find on internals cards, yet it IS AN INTERNAL card! blows my mind ;)

also supports balanced and unbal i/o as well as 'easy' i2s and very easy coax spdif i/o.

it needs a full height slot and generally is pci-only even though some new pci-e version is supposed to be out soon.

keep it in mind: if you find yourself needing to test audio amps, preamps, dacs, etc - the ESI julia card is about as good as it gets for under $1k or even higher. amazing for audio guys. stupid for gamers but we are not talking about gaming at all.

Re:Distance from the power supply

[epyT-R]

or better electrical filtering between the analog circuits and bus noise... if one uses the spdif out, that's no longer an issue if the digital stream from the application is passed unmolested to it, which isn't always a given. things like bad resampling algos in drivers used to overcome single rate DACS (48khz is most common), or analog mixing using cheap components (rare, but happens occasionally on really cheap onboard sound solutions) can ruin the sound..

I think the number one issue with modern onboard solutions is impedance mismatch between the output op amps and whatever they're driving. Many people mistakenly plug headphones into these thinking they have the current to drive them. This varies depending on the impedance of the phones, which vary greatly from model to model, but most of these onboard solutions were never designed to do this, resulting in terrible sound. In addition, many of those that have dedicated headphone outputs often suffer from the same cheap-as-possible philosophy.

Re:Distance from the power supply

[Tyler Eaves]

Almost certainly this. I've been been running a digital out from an integrated....something or other...(What ever is built into an X58 board)... via SPDIF to a proper receiver and it's, well, digital. No noise or anything.

Re:Distance from the power supply

[TheGratefulNet]

not so much Z mismatch at all, but just plain old power handling of the output amp and output voltage swing (which matters the most).

output Z is always low (low enough) on amp stages. even built in amp stages.

Re:Distance from the power supply

[TheGratefulNet]

atx supplies are switchers. switchers are HORRIBLE.

you have to do a lot of filtering to make switching supplies sound good in audio. most good audio companies go out of their way to use analog (linear) regulator style supplies.

once you get noise in at very high freq's, its very hard to kill that later. really.

one reason usb to spdfi 'wins' is that the digital power supply matters very little and there is no analog signal to care about until your stereo/amp.

Re:Distance from the power supply

[epyT-R]

Aren't those all directly related? If the voltage swing is great enough, it'll clip, meaning insufficient current, which means impedance mismatch..

Re:Distance from the power supply

[TheGratefulNet]

impedance matters in transmission lines.

this isn't one of those, though.

no matter what your load is (high-z phones or low-z phones/spkrs) you want a low output z on your amp or 'final drive' stage.

at audio freq, its not a transmission line and matching z is not a thing you do or care about.

you do want as much voltage swing as you can get. and you want to be able to source as much current as you can. those 2 mostly depend on the psu architecture and implementation.

Re:Distance from the power supply

[Smauler]

Where do graphics cards that pump audio out fit in here? I only noticed my card (gtx460) was putting out sound when I changed my monitor to my TV, didn't connect the sound, and it made me jump out of my skin.

I've since just used the dvi-hdmi cable for everything - not bothered with the motherboard sound.

Re:Distance from the power supply

[ChrisMP1]

The switching frequency is well above the limit for audio frequencies, and a good quality PSU will do plenty of filtering. USB to S/PDIF is good for a few reasons: 1) The signal is kept digital, and either differential, balanced or optical for as long as possible. This makes it hard to pick up noise. Digital-to-analog conversion should be done as late as possible because digital signals are very noise-tolerant. 2) Optical connections eliminate grounding issues (less of a problem if you obey #1, anyway). 3) S/PDIF equipment tends to be built to higher quality because it's considered somewhat "high-end" (or, at least, more than a cheap motherboard audio chip). 4) D-A conversion is done far away from the noisy CPU and data buses. It has nothing to do with the SMPS.

You should hear the sound from the vacuum tube guitar amplifier I built. The high tension (300V) is generated by a crappy 50kHz switching boost converter I designed and built myself. The noise on the supply is absolutely awful - 1Vp-p even with only a light load. You can't hear it, though, because it's 50kHz. And that's really low frequency for a SMPS.

Re:Distance from the power supply

[TheGratefulNet]

with some amps, though, they are very wideband and you don't WANT high freq garbage up there. some good phones amps go flat to 100khz and higher. they want to more than more than cover the audio range and they don't 'like' protection or LP filtering.

so, that means you have to care. if you are a source box in front of their amp, you HAVE to care. just giving you some free advice.

keep the high freq stuff out of the audio chain and the guy in the next stage will thank you.

Re:Distance from the power supply

[ChrisMP1]

Well, yes, that is true, it can be bad to drive a big amplifier with high frequency noise. Very bad, in fact - the excess load can do anything from distorting the output to causing extra heat dissipation and killing the transistors. (My guitar amplifier was meant to drive a speaker directly, and it had a low-pass filter between the gain and output stages. The tube was just for the clipping distortion sound; the output stage was a BJT class AB push-pull with a much cleaner power supply.) I was talking more in the context of this article, though - most people who own an amplifier with flat response to 100kHz are not going to be bothering with $50 sound cards, or for that matter even considering integrated audio.

Though, a couple questions (despite some vacuum tube stuff, audio is only a minor interest for me) - 1) What exactly do you mean by "don't like LP filtering"? I can't imagine how a low-pass filter could cause a problem in this case, especially if you just attenuate about 20-30kHz and up. 2) WHY do people even bother with an amplifier that has flat response to 100kHz?? That seems a bit excessive, unless you're playing music for your pet bat... Is there an advantage I'm not aware of?

Re:Distance from the power supply

[epyT-R]

Of course, you want as much headroom current as possible, but my understanding was that most output op amps lack the current to drive most sets of speakers/phones because they were really meant to drive a line-in device like amplified computer speakers. If the impedance is low enough, the amps will clip (or create distorted waveforms even before then) because of current limitations.

Re:Distance from the power supply

[epyT-R]

they act as a sound device with only a digital out. some cards include a cable to wire to your sound board's spdif out instead. There are no analog components for the system noise to interfere with (barring egregious digital noise that creates too much jitter).

Re:Distance from the power supply

[Mass Overkiller]

AC/DC is supposed to be noisy! Man, Rock 'n Roll Ain't Noise Pollution!

Re:Distance from the power supply

[Anonymous Coward]

No it is not. Most of the improvement comes from far better D/A conversion. You are basically making an assumption and it was wrong.

Re:Distance from the power supply

[Anonymous Coward]

Yes, you are missing advances from the 1990s. It turns out that humans react to sounds up to 100kHz, even though they can't "hear" them. Double blind A/B testing confirmed that people with healthy ears could tell the difference between a circuit limited at something like 25kHz and one with overtones up to 100kHz.

Re:Distance from the power supply

[Anonymous Coward]

You're talking about $120 power supplies if you want reasonable levels of noise. That's very, very uncommon.

For examples, go to hardwaresecrets and compare the seasonic power supplies with crap like thermaltake and others. Orders of magnitude difference in noise... and they're generally conservatively specced by ~30%, while thermaltake are optimists - iirc their 700w psu craps out before it gets even close to 700w.

Re:Distance from the power supply

[Anonymous Coward]

[Citation needed]. Seriously.

Re:Distance from the power supply

[darkHanzz]

The surprising thing is that digital audio introduces it's own kind of noise:
The D/A converter derives the clock from the S/PDIF interface, which requires PLL filtering for a stable clock. PLL design is surprisingly difficult, so there are systems out there where the optical S/PDIF connection introduces clock-jitter and give worse quality than a halfway decent analog connection.

Re:Distance from the power supply

[Anonymous Coward]

Very high frequencies? Really the human ear can only hear to about 40KHz, and that is hardly very high frequency by todays standards. A power supply switching at about 100KHz would be undetectable by the human ear.

Now how about linear power supplies, well they have the issue that they are horribly inefficient. On top of that the linear supplies introduce a lot of inductance into the circuit, and waste a lot of copper on the transformer. And if computer supplies were as noisy as you say then, most or all of the components attached to them would not function. The reason you need to have a power supply sit before working on it is because of the high frequency filters they place on the supply. Though this kinda makes me want to take my Oscope to my supply now.

Re:Distance from the power supply

[not flu]

Where do you get music that has frequencies above 22kHz? Is it not more likely that you would get unwanted noise than actual signal in supersonic frequencies because they cannot be deliberately mastered? Even if the high frequency signals came from the recording as intended, AND the speaker could play that high, they would just produce intermodulation distortion on the speaker degrading the quality of the actually audible spectrum.

There is individual variation of course, with claims going as far up as 25kHz. 100kHz however is clearly overkill.

Re:Distance from the power supply

[jones_supa]

I think the number one issue with modern onboard solutions is impedance mismatch between the output op amps and whatever they're driving. Many people mistakenly plug headphones into these thinking they have the current to drive them. This varies depending on the impedance of the phones, which vary greatly from model to model, but most of these onboard solutions were never designed to do this, resulting in terrible sound. In addition, many of those that have dedicated headphone outputs often suffer from the same cheap-as-possible philosophy.

This is correct. Even those basic Realtek chips are not your grandpa's noisy Sound Blaster anymore. Just get a separate headphone amplifier (it does not even have to cost much) and you'll experience the best sound improvement.

Re:Distance from the power supply

[bruce_the_loon]

One answer for your second question is that why add extra filter circuits that will attenuate the signal just because the drivers can handle the upper frequencies when you can make sure you get clean audio into the inputs and keep the amp as simple as possible.

MOSFETs can switch flat at those frequencies, but they aren't used because of the frequency range, but because they are way cleaner and efficient at audible frequencies under high power than ye olde transistors. You get the 100KHz flat range as a happy side effect.

(And yes, I know the T in MOSFET means transistor, but you know what I meant, didn't you)

Re:Distance from the power supply

[SlappyMcgee]

This all is kind of foreign to me (learning about acoustics in physics was cool and all, but not my strength). But here is a study that seems to suggest that the parent is correct: http://jn.physiology.org/content/83/6/3548.full

Re:Distance from the power supply

[FunkDup]

Most of the improvement is likely due to increased distance between the amplification circuits and the noisy AC/DC power supply.

I'm not necessarily disagreeing, but the two most important things are the converters and the clock.

Re:Distance from the power supply

[thegarbz]

they don't 'like' protection or LP filtering.

This sounds very wrong to me. Any kind of amplifier is typically pushed towards stability by adding low pass filtering. In fact the wider the gain of the amplifier the more likely it is to amplify a signal that's far enough out of phase to turn your amplifier into an oscillator. Low pass filtering is exactly what these amplifiers DO like.

Re:Distance from the power supply

[thegarbz]

They are on par with the digital outputs on the motherboard, on cheap soundcards, and on expensive gamer cards. Ultimately the benefits discussed in the articles are more to do with the analogue components and the conversion components. Your graphics card has neither.

Re:Distance from the power supply

[Lonewolf666]

In my experience, the ability to drive headphones AND the overall quality varies wildly. The integrated sound on my last 3 mainboards performed as follows:

    -Abit IC7 (Socket 478 board, bought in 2004):
Could drive headphones in terms of volume, but had horribly bad filtering. Scrolling the screen would create audible noise in the headphones. Buying a cheap Soundblaster card was quite an improvement. The Soundblaster sounds much better, and it REALLY can drive the phones ;-).

    -ASUS M2N (Socket 939, bought in 2007):
All-around OK sound with headphones. Cannot play as loud as the Soundblaster, but is good enough that I never bothered with a discrete sound card for this board

    -Asus M4A78LT or similar (can't remember the exact model number, but it is a Socket AM3 board from 2011):
OK but not really great sound with head phones at low volumes. Starts distorting the sound heavily if you turn up the volume. On this board, I added a discrete sound card again...

Re:Distance from the power supply

[gomiam]

I remember reading about russian experiments with vibrators attached directly to the skull that allowed people to hear frequencies up to 100kHz. Unfortunately, I think it was a printed magazine I read it in and I haven't been able to find it on the web.

Re:Distance from the power supply

[Anonymous Coward]

100kHz however is clearly overkill.

No it isn't. Ask audiophiles with trained ears about this. For whatever reason, relatively wide bandwidth audio amps sound better. I think this may have something to do with amps with that philosophy having short signal path designs with minimal filtering and shaping to screw with fine details. Individually those details might be difficult to perceive, but cumulatively that will affect the sound of the amp.

I have a little high quality, 10yo-plus hand-built MOSFET hifi amp from Metaxas, formerly an Australian high-end boutique audio manufacturer, that goes up past 100KHz. That company claims that it pioneered short signal path, capacitor-free amps in the marketplace. The transients and detail through good speakers, what audiophiles describe as a sense of "air around instruments", is so beautiful on good recordings of acoustic instruments. You can hear the room in which is was recorded. It is the best little hifi amp (for the price I paid) that I've ever heard and it is a collector's item.

Re:Distance from the power supply

[ewanm89]

"Rock" and "Rock And Roll" are two different styles, please see Elvis Presley, The Beatles, Buddy Holly, Danny and the Juniors, Chubby Checker and compare to AC/DC.

Re:Distance from the power supply

[TheGratefulNet]

HD audio is coming out more and more.

since I build and test DACs (and spdif switchers, too) I need to be able to play 'what is out there'.

let me tell you, I have a small (but still useful) collection of HD files (just audio) that are 88.2k (double redbook 44.1), lots and lots of 24/96k files (good point to master at) and then the 'strange' ones that are mostly about marketing with more res than needed: 176k and 192k. since my DACs have to support ALL bitrates, I need to have material to test with. you don't buy these on disc, you buy them online and they have NO DRM (so we're all happy!). they're just flac files that have 24bit audio inside and higher sample rates. I got hold of an elton john that was 24/96 and WOW was it clean. literally a copy of the master tape but dubbed over to 24/96 via a capture. any harshness that cd audio has is gone by that point. the LP record guys can stop complaining about cd sound if they listen to GOOD 24/96 audio.

and if you are not talking about cd (but blue ray or other high bit rate audio systems) there CAN be real honest audio 'up there' way over 20k. its 'felt' or sensed. how important it is, that's not for me to say. there's real honest data up there and so I don't want to chop it off or attenuate it.

Re:Distance from the power supply

[TheGratefulNet]

sorry, wrong.

compensation (comp caps) at various points keep things stable.

using an LP filter is a brute force method. very crude. can't sell $2k amps that do 'college 101' level things. you have to be better than that to be world class in audio.

the DIY designs (diyaudio , etc) are worth looking at. see how many of the truly world class (nelson pass helps out the gang from time to time, so do other famous guys) amps work. they don't filter themselves! they have bypasses where needed but they tend to have comp caps at key points to stop oscillation but STILL run flat to 100k or higher.

Re:Distance from the power supply

[TheGratefulNet]

again, that's current sinking ability of the amp section. the output-z of the amp is meaningless. the LOAD Z is very important.

simplify things: consider the source to be zero (a short) which is pretty much is, at AC. you now have a loop of a current source, a load (spkrs) and that's it. if the load is higher-z (harder to drive) you need more voltage swing to sink more current. assuming the amp can deliver whatever current its psu has, for this example.

make the spkrs lower z and you need less volts swing to get the same output. but the same amp can drive low-z and high-z just fine, as long as it can swing volts enough for the high-z.

example: my sennheiser hd580 is 600ohm or something high like that. my phones amp needs dual 30v (!) to really make an impression on those phones. otoh, my denon d2000 (very low z) can work with that same amp, just turn its vol knob down quite a bit.

none of this had anything to do with output-z of the amp. in fact, the output z of my best amp is 0.001ohms or something really stupid-low like that ;)

the key is that it uses a dual 30 psu and can swing THAT FAR into any load that lets you.

Re:Distance from the power supply

[ChrisMP1]

This gets a big ol' fucking [citation needed].

Re:Distance from the power supply

[ChrisMP1]

I'm sure it does sound good. Any "hand-built MOSFET hifi amp" from a "boutique audio manufacturer" is not going to be a piece of crap. I doubt that the fact that it goes up past 100kHz has anything to do with that sound, however.

Re:Distance from the power supply

[ChrisMP1]

As we all know, "audiophiles with trained ears" are not at all prone to misinformed bullshit regurgitation, confirmation bias, and elitism.

Re:Distance from the power supply

[nabsltd]

The D/A converter derives the clock from the S/PDIF interface, which requires PLL filtering for a stable clock.

Note that even though this problem is completely insignificant anyway, it doesn't exist at all with any packet-based, error-corrected audio like Dolby Digital.

So, if you think jitter is truly an issue, the solution is to have a sound card that can encode to Dolby Digital on the fly (Dolby Digital Live).

Re:Distance from the power supply

[Anonymous Coward]

Ok, so lets get this right...
Human hearing, at best, gets to around 20kHz (the consensus appears to be that the old 20Hz->20kHz for a young person is the norm),

ChrisMP1 designs a switching booster which operates at 50kHz, well beyond the normal human response, but you say that because some idiot designs an audio amp which *may* process signals up to 100kHz he should beware?

Ok, great, an amp which is flat to 100kHz is a mighty fine design if you want to fuck with bats, insects or dolphins, that is, assuming that you have both;

i. a device capable of reproducing source material for such an interesting audience, and
ii. source material scored for them that they might enjoy listening to in the first instance , ,

but I notice I am not alone in thinking that designing an amp with this bandwidth looks a mite stupid and somewhat 'overkill' for reproducing stuff for us mehums within the limited range we hear, remember 20kHz max on average when young, 12-15kHz at age 25, and gets worse when you're older - especially men, as our hearing deteriorates faster than that of the fairer sex.

I'll call BS on this 'sources with audio (hah) components up to 100kHz sound better', as far as I can remember when I last heard this one, this was a clutching-at-straws argument put forward by our 'golden eared' brethren in an attempt to place their BS on a quasi-scientific footing (I think the argument went, we say we can hear these differences, but if human hearing only goes up to 22kHz at best, science says we can't, ergo, science is wrong and for us to hear these 'subtleties a spectrograph would miss' the minimum top end we'd need is 100kHz, ergo we can here up to 100kHz, or something like that..I didn't pay too much attention)

I truly have to ask the question, what the fuck were the designers smoking that morning when they came up with this one? (Yes, I know, the simple answer is that they're working for companies who're out to charge $1000's for amps that fools with more money than sense will buy), to repurpose your warning about the 50kHz tone/noise, think about this, if the damn thing (amp with 100kHz flat response) starts breaking into oscillations/feeding back somewhere at a frequency above the 'normal' human range like this 50kHz, how would you know? (apart from the eventual thermal (hopefully) shutdown of the amp that would ensue, or the fried output semiconductors if the company that the designers work for want some more money out of you. If you build in 'feedback killers', then you're introducing tunable narrow notch filters into the (no longer) prisine signal pathway to your aurea aures.

Re:Distance from the power supply

[Anonymous Coward]

(-1, Pedant)

Re:Distance from the power supply

[darkHanzz]

Also dolby digital has a "self-clocking data-mode", in which case the PC is still the clock source, and the PLL story is relevant. Your comment applies when the D/A converter uses it's own clock source. Since S/PDIF has no return channel, this requires resampling of the audio, to match the (very close but slightly different) clocks of the PC and D/A converter. Just as for the PLL, there exist high-quality solutions, whether or not you particular D/A has them, can be hard to figure out, though.

Re:Distance from the power supply

[ottothecow]

So is the answer that if you are using a digital path, it doesn't matter?

My main music and video source is an HTPC (with an amd llano apu) and all of the sound and video goes into a receiver from the integrated hdmi port. My assumption has always been that as long as I am getting an unadulterated source signal (not trying to add some digital EQ to the stream or something), there would be no benefit to dedicated sound hardware. All the processing would be handled by the DACs in my receiver and thus a soundcard would be useless.

I suppose its possible that a $50 soundcard has better DACs than a low end receiver and thus using a $50 soundcard and a decent amplifier might have a better result for less money than no soundcard but a $2-300 receiver. I've been perfectly satisfied with the sound over HDMI but if I am missing out on something that a $50 soundcard could fix...someone please let me know!

Re:Distance from the power supply

[JuicyBrain]

they would just produce intermodulation distortion on the speaker degrading the quality of the actually audible spectrum.

Maybe that's what they actually heard ? With what's been said, all we know is that they could tell there was a difference...

Re:Distance from the power supply

[Anonymous Coward]

Oh be quiet! The wooden knobs on my equipment are totally helping...

Re:Distance from the power supply

[kyrio]

Yeah, all of those tracks were/are on Demonoid, et al. Many of them are fakes, especially from a certain site (studio upmixes/lazy remixes - in other words, not from the source material). I suggest you search the 'net for each of those albums to make sure they are, in fact, mixed from the original sources. You might just be running with a bit of confirmation bias/buyer's remorse.

Re:Distance from the power supply

[thegarbz]

Yes the assumption is mostly right. There are other benefits of better audio equipment such as better drivers allowing you to disable hardware mixers a more carefully designed digital path (reduces signal jitter which comes out as a harsh treble tones in extreme cases). But generally S/PDIF via the video card should be ok. Most of them I believe don't mess with the signal path as they try to pass DTS and such coding directly out to the receiver. If this is the case you won't get any benefit from an external soundcard.

Re:Distance from the power supply

[Anonymous Coward]

You forget that every circuit, not matter how powerful or weak, still takes a 50 or 60Hz signal from ground. Now I don't pretend to profess to know why, but my integrated audio literally makes a grinding, thrumming noise when my 560Ti kicks into 3d mode, and makes a slightly different frequency one when its playing through FMVs in fact the sound slightly varies as the load onscreen changes. Granted this is only possible to hear through my Studio Monitors, but it would be ther in the background of everyone headphones irritating thier ears and telling them 'this is low quality'.

That alone tells me how unshielded the audio circuit is from the other components in a PC. Add to that electrical resonance cancelling random multiples of frequencies in the circuitry and incredibly cheap and poorly made hardware, and you suddenly have many possible problems caused by integrated audio.

Discrete Audio is usually made to be resistant to all of this, in fact the majority of high end solutions for Audio are external to the case for this very reason alone. There simply is too much EM noise in the case.

Re:Distance from the power supply

[Anonymous Coward]

Okay, lets look at it scientifically. Our ears hear all the way up to around 22khz, but at 22khz they can still define between sine waves and saw/squares (cant tell much between saws and squares but can between sines and either). At 22khz on a 44.1 sample rate, the only wave possible right at the top is a saw. Because is frame 1 is at max phase (+1), frame two at minimum (-1) and frame three at +1 then you have a 22khz saw. You end up getting aliasing at the top end ultimately.

96khz, while by no means the best, gives twice as much detail at the top, allowing you to create much smoother waveforms at the top and as a result enjoy a much higher clarity and lower distortion near the nyquist.

So, scientifically, 96k is not only better, but quite easily perceivably better. And its odd you's say that we cant feel harmonies outside 22khz, because they interact with harmonies lower. You can look it up, but its widely agreed on, that simply recording the original TB808 drum machine (which made sound up to 36khz and more) up to only 22khz made it sound flat and bland.

Re:Distance from the power supply

[Anonymous Coward]

Audio Amps usually have a separate power source.

Re:Distance from the power supply

[Anonymous Coward]

I'm sure it does sound good. Any "hand-built MOSFET hifi amp" from a "boutique audio manufacturer" is not going to be a piece of crap. I doubt that the fact that it goes up past 100kHz has anything to do with that sound, however.

#40956551 here: note I said ".. for whatever reason ....". I wasn't saying that wide bandwidth is responsible alone for that liquid MAS sound; sound is a systemic quality. But wide bandwidth in this case is part of a particular holistic design philosophy that works. There are many other factors, Kostas Metaxas hates capacitors and has very specific ideas about power supplies, for example. For more info http://www.metaxas.com/pages/masnewfiles/index.html

Metaxas stuff is great, a great pity it seems to be out of production. You can pick some of it up on eBay sometimes. I met Kostas Metaxas once in Melbourne and have exchanged a few emails in the past, an interesting guy. His best sales were providing top of the line systems to the very rich eg oil princes in the Middle East. Service could be a problem since his bigger amps were arguably not that reliable. For those in Europe, Kostas did recommend a guy in Germany who will service it.

Re:Distance from the power supply

[wdef]

As we all know, "audiophiles with trained ears" are not at all prone to misinformed bullshit regurgitation, confirmation bias, and elitism.

Yawn. This is the stock straw man for derailing any discussion about audio. There are real differences that can be described and heard also.

Re:Distance from the power supply

[wdef]

Okay, lets look at it scientifically. Our ears hear all the way up to around 22khz, but at 22khz they can still define between sine waves and saw/squares (cant tell much between saws and squares but can between sines and either). At 22khz on a 44.1 sample rate, the only wave possible right at the top is a saw. Because is frame 1 is at max phase (+1), frame two at minimum (-1) and frame three at +1 then you have a 22khz saw. You end up getting aliasing at the top end ultimately.

96khz, while by no means the best, gives twice as much detail at the top, allowing you to create much smoother waveforms at the top and as a result enjoy a much higher clarity and lower distortion near the nyquist.

So, scientifically, 96k is not only better, but quite easily perceivably better. And its odd you's say that we cant feel harmonies outside 22khz, because they interact with harmonies lower. You can look it up, but its widely agreed on, that simply recording the original TB808 drum machine (which made sound up to 36khz and more) up to only 22khz made it sound flat and bland.

Interesting post.

Re:Distance from the power supply

[wdef]

Filtering to limit upper input bandwidth introduces artefacts. The less in the way of the signal to interfere, the better quality of amplifier.

Re:Distance from the power supply

[harrkev]

That quote is a complete load of bull.

I am in my low-fourties (starting to get up there in age). I cannot even HEAR anything above a certain frequency (around 14 KHz as I recall). My kids can go up higher. Do you think that I can tell the difference between two sounds that I cannot even hear? I went to a few concerts as a youth, and I used to turn up the stereo in the car, but I am hardly unusual in that respect. I consider my hearing to be at least typical for a person my age.

If a person claims to be able to hear 22 KHz, then they are either still in diapers, or as full of crap as a diaper is.

Re:Distance from the power supply

[jaymemaurice]

I bought an Asus XONAR Essesence STX because I listen with headphones often. Best purchase ever. For headphones, you can notice the clairity difference between headphones plugged into the reciever (connected by S/PDIF) and the analog headphone port. There is NO background noise on the head phones and it can drive headphones with unreasonable power at instant permenant hearing damage levels.

For digital out, it allows the card to send a Dolby Digital signal out over the S/PDIF - which is of huge value in that it stops my reciever from switching in and out of dolby digital when I pop in a DVD. Also, I don't have to mess with the controls on the reciever. I can choose the headphone or digital out from software and leave everything connected.

Unfortunately, I don't think you can get the STX for $50 :(

Re:Distance from the power supply

[lars_stefan_axelsson]

Okay, lets look at it scientifically. Our ears hear all the way up to around 22khz, but at 22khz they can still define between sine waves and saw/squares (cant tell much between saws and squares but can between sines and either). At 22khz on a 44.1 sample rate, the only wave possible right at the top is a saw. Because is frame 1 is at max phase (+1), frame two at minimum (-1) and frame three at +1 then you have a 22khz saw. You end up getting aliasing at the top end ultimately.

Well scientifically that's not true. While everybody and his brother seems to have heard about the Nyqvist critiera, very few seem to know how you actually *recreate* the signal after sampling.

And you don't play "connect the points" as you seem to suggest, but filter your time discrete sampled signal with a filter with a transfer function that looks like sin(x)/x (http://en.wikipedia.org/wiki/Nyqvist_limit)

Nyqvist in fact says that a signal sampled at twice the maximum frequency can be perfectly recreated up to that limit. Note perfectly. That means that the wave forms will be correct, no sin waves converted to sawtooths or whatever.

That you percieve sound as better or worse sounding depends on a lot of other things than the Nyqvist limit...

You don't say...

[zonker]

$50 sound card produces better audio than a 50 cent onboard chip.... You don't say.

Re:You don't say...

[MrL0G1C]

Actually their RightMark audio analysis's don't show this at all, frequency response, THD, noise, are all so close between devices that a human wouldn't be able to tell the difference between them, the ignoramuses at TechReport however don't know how to read the graphs/understand the limit of human hearing and came to erroneous conclusions.

Their section on different peoples opinion of the various audio devices does not state the result of the blind listening test and so is useless, why even bother with tests that are not blind?

Re:You don't say...

[larppaxyz]

And when using just digital output, there is no difference at all anymore.

Re:You don't say...

[gazbo]

I think what's most telling about their analysis is on the 96kHz plots. The Realtek consistently drops to nothing around 20kHz, and yet apparently that didn't mean anything to them other than "look how well these results fit with out hypothesis". Anyone who actually knew something about didgital audio would think "either I've set this up wrong or the drivers/hardware are bust, because this thing is blatantly stuck at 44.1kHz".

The only other thing to be gleaned from the graphs is that running at 96kHz is pointless because the supposedly better cards' performance FUCKING SUCKS past 20kHz.

Re:You don't say...

[makomk]

Anyone who actually knew something about didgital audio would think "either I've set this up wrong or the drivers/hardware are bust, because this thing is blatantly stuck at 44.1kHz".

Or it's just got a low-pass filter with the cutoff set at 20 kHz which can't be disabled. You need one for proper signal reconstruction at the 44.1kHz sample rate, and it's not like most people are going to notice that their onboard sound can't actually output frequencies above 20 kHz in its 96kHz sampling mode.

Re:You don't say...

[gazbo]

I think claiming to have 192kHz DACs and then sticking a 20kHz filter in front of them would be...misleading. And looking at the datasheet it certainly suggests that the cutoff moves with sampling rate.

But whatever the cause, the point is that something like that should not pass without comment; that it has done indicates to me that the reviewer may not be particularly familiar with the subject.

Re:You don't say...

[gazbo]

"didgital"? WTF am I typing?

Re:You don't say...

[Anonymous Coward]

Considering around 18KHz to 20KHz is the typical cut-off for normal human hearing. Obviously some can hear up to a slightly higher KHz while others can hear 18KHz

Re:You don't say...

[Anonymous Coward]

The unit is incorrect. Units are important in science to precisely stating the specs if used correctly. It should be 192ksps where it is a sample per seconds, not frequency which can be confused with the actual Frequency Bandwidth at 20kHz.

A 44ksps would barely reproduce a sine wave at 20kHz as you only have 2.2 data points per cycle!! Try draw a waveform like that and claim it is any good. All the samples in between are interpolated with a smoothing function. As the sample rate is increased, you'll get more of the "real" data filling in the gaps. That's why we use oscilloscopes that are at least a few time higher sample rates than the waveforms we are measuring.

Re:You don't say...

[Anonymous Coward]

20kHz is the generally accepted upper limit for the audible range, so it's a reasonable place to put the filter cutoff frequency. There is still a good reason to use high sample rates. With a sample rate of 44.1kHz, the Nyquist rate is 22.05kHz. If the analog filter rolls off at 20kHz, a significant portion of the signal above the Nyquist rate will get through, leading to poor reconstruction. (Each pole in the filter decreases gain by 20dB/decade for frequencies above the pole, and 22.05 is only .07 decades above 20) If the sample rate is 192kHz, then the Nyquist rate is 96kHz. That's 0.68 decades above the filter cutoff frequency. Using the higher sample rate will lead to a significantly better reconstruction when the same analog filter is used. This is the concept of "oversampling" that used to be used in marketing CD players 20 years ago. In practice, oversampling is a way to get better performance out of a cheaper analog filter. It is generally cheaper to increase the speed of the digital electronics than to increase the number of poles in the analog filter.

Re:You don't say...

[Anonymous Coward]

I totally agree with you anne. I read the article yesterday and it's basically a sales pitch.

Anyone not happy with free onboard audio won't be pleased by $50 offboard audio. Period.

Re:You don't say...

[Anonymous Coward]

No

No, no, no, no, no.

Soundcards work with PCM. Because of the _definition_ of PCM (go on, look it up) you can define the wave perfectly with as little as one sample per peak. In practice the limits of our technology mean you can't quite manage that, but you can get damn close, easily better than 20kHz with a 44.1kHz sample rate. On input the low pass filter cuts out everything else so the sampling will work correctly done this way. On output the reconstruction filters turn the impulses into a smooth curve within this frequency range.

You will see a lot of _idiots_ writing software that represents 44.1 kHz PCM data as a square wave. This means they know nothing whatsoever about PCM audio and you should treat it the same way you'd treat someone who assures you that the reversing lights on a car are its headlamps. In fact the samples are _impulses_ and should be represented either as a series of unconnected vertical lines or, since it's probably easier for humans to understand, as a continuous smooth sinusoidal curve.

You mentioned oscilloscopes, I guess you have one. Connect one to any old 44.1kHz soundcard. Play some music. Look at the lovely smooth curves on the oscilloscope. Voila.

Re:You don't say...

[theshowmecanuck]

I think most people can't tell the difference. I think the only real correlation is that the higher the audiophile pomposity index a person has, the more they think they can tell the difference, whether they can or can't... see the first sentence of this post.

Re:You don't say...

[antdude]

That explains why its low frequencies suck. My Creative SB Audigy 2 ZS and Live! sound cards (mainly for gaming and their hardware EAX -- don't game anymore though!) did better than this onboard RealTek audio. However, I have yet to find a good cheap PCIe sound card for both Linux and Windows XP Pro. SP3. I only listen to MP3s, watch videos (DVDs, HDTV OTA feeds/recordings, streaming videos, etc.), quick and random Flash games (rarely computer games), etc.

Discrete sound

[ozduo]

Is farting silently

Re:Discrete sound

[Gadget_Guy]

Discrete sound...Is farting silently

No, that is a soundless indiscretion.

Re:Discrete sound

[TheGratefulNet]

aka, 'the sound of one butt clapping'.

or, well, something like that.

usb to spdif, then to your home stereo

[TheGratefulNet]

there, done in one.

cheap cmedia usb sound dongles (not all dongles are cmedia, in fact most are not so you have to shop carefully) and also the burr brown PCM series all do a decent job of converting 44 and 48k audio (including dvd audio downmixed to 2.0) to spdif.

everyone's avr, today, has opto in. the sound card dongles send out usb audio over opto to spdif-in of your home stereo. if its 5.1 or newer, it will accept opto just fine. (aka toslink).

nothing else to care about, pretty much. let your stereo (or DAC) do the heavy lifting. usb audio is the way to go (for future, use UAC2, usb audio class 2 which works fine with linux and some hacks on windows at 24bit and 192k, personally verified to be bit-perfect).

Re:usb to spdif, then to your home stereo

[Tyler Eaves]

Going to an SP/DIF connection there shouldn't be any conversion at all - a DAC isn't in the circuit). It's pushing raw PCM data over the wire.

Re:usb to spdif, then to your home stereo

[TheGratefulNet]

in fact, its bit-perfect since there's no (almost never) a hardware mixer for usb audio. avoid software mixers and you always get the full raw non-scaled audio stream out of usb.

if you do select hardware mixer, the control will move but not change anything. which is good!

Re:usb to spdif, then to your home stereo

[Tyler Eaves]

I remember when I obsessed over such things. These days I mostly don't worry about it and spin vinyl. Sounds better anyway.

Re:usb to spdif, then to your home stereo

[epyT-R]

depends though.. some onboard sound drivers are known for crappy output.. they resample unnecessarily and/or they do it badly.

Re:usb to spdif, then to your home stereo

[TheGratefulNet]

these days, i'm not sure ANY onboard sound forces a resample.

soundcards 'offer' freqs to the sound system. back in the dinosaur days, they would offer only 48k and the o/s would have to resample.

but xp and win7 all deal with 44/48 split just fine. linux always has.

onboard audio (if its native spdif via jacks or headers) is usually bit perfect. its there to give 5.1 and even 7.1 digital out. the days of speaking 'only 48' ended 5yrs ago or more (I forget). a long time, at any rate (lol, any rate!)

Re:usb to spdif, then to your home stereo

[Anonymous Coward]

Or hdmi from the graphics card to the receiver (nvidia/ati and even intel onboard these days support hdmi audio). Toslink is limited to either 2 channel PCM or compressed DTS/AC3. You could encode 5.1 on the fly to DTS, but with HDMI you can just stream multi-channel PCM.

Re:usb to spdif, then to your home stereo

[epyT-R]

I'm willing to bet they do, but it's hidden behind a software resample, at least for the dacs behind the analog outs.. I would hope that modern drivers do not resample for the spdif unless it's not a supported frequency/bitrate. I'm talking about the PCM standard here, not the raw output used for ac3/dts.

Re:usb to spdif, then to your home stereo

[adolf]

Meh. I doubt many consumer systems have bit-perfect outputs, these days -- it seems there's always something in the loop to screw things up (including volume controls, whether implemented in software or otherwise).

If it actually is a bit-perfect output, then you can take a DTS CD*, play it on the computer over S/PDIF to any modern(ish) AVR, and very simply get 5.1 channels of analog output without additional fuckery. It's a fun test, and I think you'll be surprised at the result: Chances are good that your "bit-perfect" S/PDIF out fails miserably.

*: Yes, these exist. They're fairly neat.

Re:usb to spdif, then to your home stereo

[thegarbz]

Why USB?

Every motherboard I've ever owned or bought has had S/PDIF outputs available. My previous motherboard did so via an onboard RCA, and my current one via a set of headers that I've seen on every motherboard made in the last 5+ years. Personally I just run from this header to a socket on back of the computer, via a 4m long cable to a DAC. Sound is perfect.

Re:usb to spdif, then to your home stereo

[wonkey_monkey]

I don't know about SPDIF playback, but I do know that I've never found a way to do a bit-perfect SPDIF record on Windows 7 - something, as a few people have now said, always gets in the way. On Windows XP, on the other hand, it is possible.

Re:usb to spdif, then to your home stereo

[adolf]

Yep.

Back in the day, I had a Zoltrix Nightingale card (actually, I still have it -- all $23 worth of it including the optional TOSLINK module), and was able to do verifiably-correct recording and playback over S/PDIF under Linux and Windows.

The C-Media 8738 chip on that card is simultaneously marvelous in its simplicity and signal routing and lousy for its analog audio quality. As an S/PDIF IO, though, it did just fine. (I bought it specifically to use with an external DAC.)

A bit further back, I was involved with a small studio that had a Lexicon Core2 card which would do exactly as it was told (even if it was wrong), and it also made accurate digital recordings -- 10 channels at a time. That was fun when it worked right, though it was much less-than-fun when the owner recorded a bunch of things incorrectly and I had to patiently explain to him (again) that he and his recordings were fucked.

These days? Meh. It's all blurry, apparently by design. I don't much like it from the purist standpoint, but as a realist I hear (with my own ears) that it's not so bad. Gone, at least, are the ticks-and-pops from mismatched clocks... and we seem to be making up for yesteryear's inefficiency with modern overkill horsepower and vastly improved analog electronics at all levels (including on motherboards -- those little ICs keep getting better).

My X-Fi is said to have one of the best arbitrary sample rate converters around, for instance, though I also understand that the majority of the millions of transistors on the (heatsinked) DSP are dedicated to that task alone. I've done useful and (at the time) modern things with computers that, in total, had fewer transistors than are used for this solitary purpose.

Re:usb to spdif, then to your home stereo

[fa2k]

While it's true that it's "bit perfect", there can still be jitter if the clock on the sound card and the clock on the receiver aren't in sync.

Re:usb to spdif, then to your home stereo

[TheGratefulNet]

the new audio class (UAC2) for usb totally solves that.

its so cool. yes, its exciting to folks like me.

instead of a push model where the sound card gets TOLD what samples to drop (on line-out) and when; the sound card now PULLS data on its own schedule ("I'm hungry now, feed me!") and all timing - ALL - is created locally from very high end xtal clocks. this is FINALLY where 'timing wont matter' on the computer. it can be off by half a bit width, randomly! it won't matter as long as the receiver (dac) has a deep fifo, can construct its own stable clock and keep the data flowing with no overs or unders on the computer side.

uac1 is limited to 16bit/48k on usb1. usb2 audio has always been bulk-based (like file xfer) and proprietary. with the recent linux kernel and things like 'audio widget' (see diyaudio for that), you can finally run UAC2 class audio and KNOW that your computer timing is 100% irrelevant to the output sound.

expect to see more UAC2 boxes over the next year. this is what you want. it will make uac1 into 'toys'; and uac2 will be where all the home theater and audiophile guys will want to be.

Re:usb to spdif, then to your home stereo

[TheGratefulNet]

again, the chips have been able to support literal 44.1/16 for well over 5 years now.

the push to dd5.1 and dts FORCED them to drop the 'whole world is 48k based' view.

really, the chips are fine.

software layers suck but that's NOT the motherboard or chip's problem.

run linux on almost any modern mobo and I bet I could get 44.1 bit perfect to come out with little trouble.

win7 still messes with things, but still not too badly. asio fixes that, for those guys.

Re:usb to spdif, then to your home stereo

[TheGratefulNet]

when zoltrix sold out, I bought 3 or 4 of those cmi8738 cards. they were great! I have a few spares, still, but don't need them. usb is doing my audio now.

some guy on google code also did a PROPER cmedia driver for xp. it was great! really a good driver. not sure if its there for win7 or not.

win7 is doable. I tend to avoid it since I know how to make xp work in BP mode. for rmaa testing, we need that. win7 makes it harder and its still not as 'good' as xp was in doing audio and using all the work-arounds.

Re:usb to spdif, then to your home stereo

[TheGratefulNet]

onboard spdif can be fine.

sometimes, though, I've seen noise riding on the spdif. and if you don't use trafos to block (pulse brand or similar) you risk having NASTY ground loops.

if you use opto, you avoid all that. but still, noise on the psu line (optos tend to be 3.3v or 5v based and they have 3 wires: gnd, power and data-in) can cause data errors on the receiver IF the receiver is not great at rejecting 'junk'. some do better than others. some pass the noise along! sometimes you get screeching noises even when the spdif TX is not talking.

usb spdif interfaces tend to be slightly better and the usb audio format *generally* disables hardware mixer controls so you can't ruin the bitstream even if you try using a volume slider.

onboard chips MAY have a hardware mixer and that's bad.

Re:usb to spdif, then to your home stereo

[BigPhatPhuck]

Na, you're just getting old enough that you can't hear the difference ;)

Re:usb to spdif, then to your home stereo

[Anonymous Coward]

Ya because I want my PC audio coming from 30 feet to my right with the center sound stage squarely in my right ear. No thanks. My PC isn't in my home theater listening position. I am not going to run a separate zone just for my PC either.

Re:usb to spdif, then to your home stereo

[adolf]

The chips were able to support literal 44.1/16 20-ish years ago, too: In a world before AC97, sound cards tended to do one thing at a time, and do it exactly as they were told to do so, whether the rate was 44100Hz, 22050Hz, 48000Hz, or 11025Hz.

That said: Software layers do suck, and that's the system's problem. I don't normally use a sound card, per se, but a system that contains one which also has at least one layer of software in front of it.

I've done bit-perfect things under Linux and FreeBSD using paid-for drivers from 4front, with good results. But the world today (with layers of ALSA and/or PulseAudio and/or ESD) is dramatically more...vague. Certainly it can be made to work, but you're likely to temporarily break everything else in the process, which in turn results in a very poorly-performing system.

If I was spending $50

[slackware 3.6]

it would be on a cheap video card before a sound card. I never bought any sound cards after they started puting them on the mobo. Sure the sound could be better but I have a stereo for playing tunes and if i'm playing games at night I'm using headphones anyway. A better soundcard is a non-issue for most users.

Re:If I was spending $50

[phluid61]

Except that I can still hear my mouse through every on-board sound system I've used recently.

Re:If I was spending $50

[Anonymous Coward]

then you probably have a ground problem with your house wiring

Re:If I was spending $50

[TheGratefulNet]

that's a good reason to go digital. including toslink (fiber opto). no ground or even power connections between your stereo (or set of spkrs and an amp) and the pc. that side gets solved easily.

let the pc be as noisy as it wants. square waves galore! wheee! and it should not affect the sound system.

go with some card that gives toslink fiber out. then bring that to your stereo. you should no longer hear any 'clocking' junk.

Re:If I was spending $50

[Smauler]

Many video cards now put audio out too - I've got an old DVI to HDMI cable from the back of mine, and it's spitting out audio happily.

Re:If I was spending $50

[Mashiki]

Seriously? A good soundcard will last you across multiple generations of hardware. Heck my old Soundblaster Live lasted from 2002 right up until 2011 when the ports on the back finally failed, and I simply gave up on trying to solder in new ones. You know for 8 years for a PCI card that cost me $80, that's a pretty good investment. With the new PCIe jobs? Same deal. Though I have an Asus Xonar DG(picked it up on sale for $29 last year from newegg), I should easily get 5-6 years from it.

I figure sometime next year I may get a new PCIe soundcard to replace this one if I build a new rig. Skimping on a soundcard is silly, especially when you can get them on sale.

Re:If I was spending $50

[slackware 3.6]

Those big plexi glass windows some people have (i got a box that has one to) really suck is you have any kind of radio transmitting in the house. Has something to do with the steel case shielding your box and also preventing your box from interfering with other devices and you computer is grounded through the power supply so AC has a point.

Re:If I was spending $50

[TaoPhoenix]

I'll reply to you.

In Personal Computing in the true sense of Personal and not corp-crap or netbooks or whatever, if you're gonna buy a goddamn comp, spend a few hundred bucks to do it right. No one except the media wins with these "budget parts" stories.

So forget the $50 sound card. What can you get with $80?
Forget the $50 video card. What can you get with $80?
Spend an extra $20 on the fan. Spend an extra $20 on a key cable. Spend an extra $60 on a better HD that has capacity to better meet your growth.
Spend an extra $40 on a better casing. I added a special extra chip for data conversion like Audacity Sound Processing 30% faster.

So yes, it adds a few bucks cumulatively. But you help mitigate stupid "cheap crap failures" that risks years of useful life. 6 years later into my op life I think I have a medium grade HD problem, but it's still ticking, so I have some time. Not like a total crash yet when some dumb part blows the whole board.

Re:If I was spending $50

[slackware 3.6]

I still have my Soundblaster Live its in a box with my Diamond Monster II just because they were so cool and in case my onboard sound craps out but i don't want to use it in case I break it (to many memories of Quake related sounds from that SC).

Re:If I was spending $50

[Mashiki]

Heck, any videocard that runs over $170(really anything in the previous generation and marked down now that the nvidia 6xx and ati 7xxx series are out) have a HDMI connector on them just for that. The only real complaint most people have about them? The audio can be a real pain in the ass to get working, once it's working not a problem.

Re:If I was spending $50

[lowlymarine]

The problem is that if you spend $700 on a video card today, you will get the same Vsync-capped performance you would have gotten out of a $350 card, and six years from now, as you suggest it should last, you would have a brick that can't handle anything remotely modern while the hypothetical other guy would only have a three-year-old $300 card that beats your $700 six-year-old card into the ground.

Six years ago, the Core 2 Duo X6800 and GeForce 7950GX2 were the top-of-the-line parts, costing a grand and $700, respectively. Within two years, both were getting clobbered by parts that cost half as much. Today that $1700 combination wouldn't even be competitive with a $75 A6-3650.

Re:If I was spending $50

[antdude]

How good are those video cards' audio compared to onboard and sound cards?

Re:If I was spending $50

[Anonymous Coward]

"That" has nothing to do with "digital" and has everything to do with circuit isolation.

Re:If I was spending $50

[twosat]

I discovered this "feature" a few years ago when I bought a computer with an integrated sound card. It was quite noticeable, but I found that by turning up the master volume and turning down the volume knob on the speakers it ceased to be a problem.

Re:If I was spending $50

[Mashiki]

Depends on what they're using for a coprocessor but in most cases they're on, or above quality to most of the Yamaha stuff.

Re:If I was spending $50

[wierd_w]

It was my experience (Especially with nasty horrible SoundMAX integrated audio before the integrated HDA bus audio) that the audio channel would pick up AM radio frequencies produced by various components inside the chipset itself, and produce funny "Birdie" noise when the processor was doing things, and especially when the PCI bus was pushing a lot of bits. This caused serious issues when playing video games. The only solution I found was to disable integrated audio, and switch to a cheap CMedia PCI card.

Worked like a charm.

Re:If I was spending $50

[Anonymous Coward]

Six is still pretty new. I just spent the evening installing Xubuntu 12.04 on an original 1.2GHz Thunderbird Athlon from 2000. It took forever coming off of DVD, the motherboard is so old it doesn't support USB booting. It makes a fine MythTV box for playing back music in the garage and light web browsing.

Re:If I was spending $50

[hackertourist]

Spend an extra $60 on a better HD that has capacity to better meet your growth.

No. Buy a HD that is large enough for your current situation + 1 year, then spend the $ 60 next year (or whenever you run out of space) to buy a HD that's much larger than you can buy now.

Also, 6 years is a long time for a harddisk. Based on past experience I've started to replace the HD preventatively every 3-4 years.

Re:If I was spending $50

[flimflammer]

Yup. Discovered that by accident when I got a new TV and hooked it up to my machine through DVI -> HDMI. Suddenly Windows was telling me I had a new output device. Checked the device list and lo and behold there was a TV icon and all my Windows sounds were coming through it.

Re:If I was spending $50

[Luckyo]

I want to chime in here. I am still using SB Audigy2 that I got back in 2000. I've changed computers three times during this time period. It outlived all of them and still works wonderfully. The way it works, I wouldn't be surprised to still be using it in ten years.
My parents have my hand-downs and they're still using my older system that is using an even older SB Live! card. That one is still alive and works just as well.

Worth noting that onboard audio is pretty terrible for me because I have Logitech Z-5500 set hooked in, and of all onboard audio I tried, all of them sound terrible on it.

Re:If I was spending $50

[TaoPhoenix]

Well, I thought ahead and had part of a situation of "best finance I'll have for a long time". So I spent actually a fair more on HD's to "fire up and forget". So I have a 1.75 Terabyte combined storage and space wise it will last me forever. It was a pocket in time where those were first possible, plus the first Quad Core Kentsfield designs, when the first Vista disasters were hitting full steam, that I wanted a computer I could just sit on and ride out theoretically to the emergence of what is now probably Windows 9.

Re:If I was spending $50

[antdude]

OK, how are they compared to SB Audigy 2ZS sound card for listening to audio with music, computer games, videos ( DVD, HDTV recordings, streaming videos, etc.)?

Re:If I was spending $50

[CronoCloud]

Heck, any videocard that runs over $170(really anything in the previous generation and marked down now that the nvidia 6xx and ati 7xxx series are out) have a HDMI connector on them just for that.

Even the cheap GT2xx series has it.

Re:If I was spending $50

[Bert64]

I used my first gen soundblaster awe32 (with upgraded ram) from around the time it came out, up until i retired my last machine that had isa slots...

Re:If I was spending $50

[Mashiki]

Couldn't really tell you on that one Ant. I've never owned a 2ZS to test against. But I'm guessing you could probably give it a try if you're really bored and have the hardware lying around, considering most yamaha audio processors are considered above average to excellent quality, depending on the card and co-processor setup. It should come out above the 2ZS, especially since you're looking at HDMI.

Re:If I was spending $50

[antdude]

Wait, I have to use HDMI to my speakers? My speakers use the old fashion regular audio cables. :(

Re:If I was spending $50

[gl4ss]

Wait, I have to use HDMI to my speakers? My speakers use the old fashion regular audio cables. :(

most home 5.1+ amps take hdmi nowadays, and pass the hdmi through and take the audio.
alternatively most displays, even some very cheap, forward the spdif audio from the hdmi cable.

the real reason why the hdmi audio quality is good is that since it's all digital it's just how good the dac's in your amp are which matters.

Re:If I was spending $50

[gl4ss]

there's usually no difference between 50 and 80 bucks parts anyhow. funny thing that.

Re:If I was spending $50

[antdude]

Ah, I don't have a fancy sound system. Just a basic 2.1 analog speaker setup (Logitech Z-2300).

Re:If I was spending $50

[jaymemaurice]

You just aren't amplifying the noise anymore... the noise is still there.

Re:If I was spending $50

[jaymemaurice]

Surely you jest - every motherboad I had from that era I have replaced capacitors on, then binned because it was still too slow.

Sound Blaster is dead

[Anonymous Coward]

I dumped my Sound Blaster Live! 5.1 for Realtek ALC888 and never looked back. I never heard such clarity until I went with this onboard chipset.

Also, I tried to compare a $99 Razer Barracuda AC-1 with the ALC888, all I could make out is that the back channels get more bass than the front. I'm guessing there's a placebo effect in play here with the little DSP cards...

Re:Sound Blaster is dead

[Anonymous Coward]

The Realtek may be better than the Soundblaster Live!, but I still have an Audigy 2ZS Platinum card in my PC simply because it is, in my opinion, much better than any on-board sound chip to date.

Re:Sound Blaster is dead

[Anonymous Coward]

As a frer game developer I'll never voluntarily give money to Creative. TFA points out that Creative's environmental audio didn;t get support after 2008. Want to know why? As soon as the hardware could support shadow maps, Creative could no longer use the 'Carmack reverse' software patent as a bludgeon to get game developers to support their crap. Good riddance.

Re:Sound Blaster is dead

[flimflammer]

If only Creative wouldn't have shit in their customers laps when Vista rolled around and actually offered proper drivers for the thing. Also have one of those cards, but I need to use hacked drivers to get any of the features the damn thing is supposed to provide. I mean Creative's drivers don't even support surround sound for this card beyond Windows XP. I swore I'd never buy another Creative card after that, which ended a long tradition of buying Creative sound cards.

Re:Sound Blaster is dead

[Luckyo]

Worth noting that "hacked" drivers are based on real drivers. They only offer a different front end that can access features that exist in drivers but removed from front end.

Unless you're talking about something I don't know, which is unlikely as I spent days researching driver issues for my audigy2 and win7.

Re:Sound Blaster is dead

[antdude]

Agreed. My old PCI Audigy 2 ZS sound card was better (bass/low frequencies, clarity, etc. -- not an audiophile and only have a 2.1 Logitech Z-2300 analog speakers!) than my onboard RealTek audio. I have yet to find a replacement for it even though I don't game a lot for its EAX these days. I still listen to MP3s, watch videos (including HDTV recordings), movies, etc. I wished newer motherboards had PCI slots to reuse it since it still works. Yes, Creative drivers suck but the card is better than onboard audio! :(

Re:Sound Blaster is dead

[jaymemaurice]

Yeah you and me both. I owned everything from the 8-bit sound blaster (that had the proprietary CDROM interface) to the high end Audigies but that all went to shit with the SB Live and Audigy

I actually thought about giving Creative one more try and bought an Audigy 2 ZS, but it and it's drivers sucked. I traded it in (paid a hefty restocking fee) and paid extra the Asus Xonar STX.

They HAD me as a loyal customer, but now I own nothing of theirs.

Given that Realtek et al can't make a good chipset

[sethstorm]

I'm not surprised that the integrated chipsets (usually Realtek) get beat by even $50 hardware. They're usually from companies that can't sell a chipset of any type unless it is included with a manufacturer due to not being able to make any provision for performance.

Those companies couldn't make a good chip to save their lives. Or even an acceptable one.

$50? Try a $250.

[X0563511]

I have this for studio purposes, but this thing sounds beautiful.

If I chose to, 96khz 24-bit. 2-in, 2-out, SPDIF support if I chose to use it. (technically 4 in 4 out, but that's mono.)

Re:$50? Try a $250.

[Anonymous Coward]

I have a $20 sound blaster audigy SE that can do 96 KHz*24-bit in 7.1 and sound beautiful.

Re:$50? Try a $250.

[Anonymous Coward]

Sorry, the soundblaster is a nice general-purpose card but is not very well fitting in a studio environment. It has no (very) decent ASIO drivers, and the latency is not that great. It is doable, but for real quality mixes you need a better card.. really. You see - the Soundblaster cards are nice, but have build-in filtering for game purposes (EAX to name something). This gets in the way if you want "clean" untreated sound - witch is extremely important if you want to make a good balanced mix.

There are some good (relatively) cheap cards like the M-audio Audiophile 2496 or the ESI Maya or Juli@, or external ones like the M-Audio FastTrack or Focusrite VRM box (good for virtual speaker systems). They are a bit more expensive than the Soundblaster, but are great for studio purposes.

At the other hand - a studio card is not well fitted for gaming purposes (no EAX), so you can use the on-bord card for that (most games do rely more on effects than on sound quality).

Re:$50? Try a $250.

[Anonymous Coward]

for real quality mixes you need a better card.. really.

Not really.
Latency only matters when recording audio, and even that not in all cases. It doesn't matter at all for mixing.
For quality mixes you need skills far more than anything else. I'd rather have my mix done by a skilled guy with the crappiest soundcard than to a noob with the best gear.

You see - the Soundblaster cards are nice, but have build-in filtering for game purposes (EAX to name something). This gets in the way if you want "clean" untreated sound - witch is extremely important if you want to make a good balanced mix.

No.
EAX is only used in _some_ games. And all those additional features in "gaming" soundcards are _optional_.
For listening to music or mixing, you get the untreated sound, just like with every soundcard.

Re:$50? Try a $250.

[Anonymous Coward]

You realize that games have to specifically target EAX right? Otherwise there is no such filtering right off the bat.

Re:$50? Try a $250.

[Luckyo]

You can force many EAX features in drivers, but they will go system wide.

Re:$50? Try a $250.

[Anonymous Coward]

I have one of these: http://centrance.com/products/dacport/ which is also not sub-$50. But it is awesome.

Re:$50? Try a $250.

[X0563511]

I've got a Sound Blaster myself. Specifically an Audigy 2 ZS - which was the last one that featured discreet DSPs, and used decent quality DACs. Models after that part dropped the DSP (in effect a "winmodem" for sound) and switched over to the same crappy DACs that most on-board sound uses.

So, in effect I have the two sound cards. The M-Audio I referenced earlier is fed into my speaker's Line In, with a ground-loop isolator (as otherwise I get some nasty noise when both devices are connected). This way I still get windows sound etc when I've got the M-Audio working via ASIO, and I get EAX in my games should they feature it.

Ensonic Soundscape

[Anonymous Coward]

I still have Ensonic Soundscape Elite sitting here. If only there was a motherboard with enough room (and the right bus) to plug it in ;)

I had some cheap ass PCI soundcard that had a Yamaha chipset on it about 10 years ago and it was the best sounding soundcard I ever had. I'd still be using it if there was Windows 7 drivers for it.

solder 3 wires and get good toslink out

[TheGratefulNet]

a DIY I wrote on how to open up a cm102 (cmedia usb audio dongle) and find the 3 solder pads you need to connect in your own toslink (TOTX) opto transmitter for your home stereo:

http://farm5.static.flickr.com/4086/5052505190_07d7ec5903_b.jpg

http://farm5.static.flickr.com/4148/5052506250_c71b26586a_b.jpg

it was just that simple. there was already an onboard cheap-o toslink sender but I prefer the standard square block style.

the TOTX part is a dollar or so at digikey or mouser (suppliers). the usb dongle is $15 or less, often much less. make sure its cmedia and cm102. it will work very well then.

usb powers it and you know its working when you get the red light out of the toslink end ;)

I'm not sure it passes dts or dolby digital but its fine for 44.1 cd audio (and mp3) as well as 48k dvd downmix to stereo 2.0

Re:solder 3 wires and get good toslink out

[thegarbz]

Why go optical? In terms of digital transmission the conversion from electrical to optical and back introduces possible jitter. While some of this is likely audiophile wank there's a measurable reduce in signal jitter by omitting the optical component and going straight down a short run of cable. Optical does help with potential grounding issues though which may arise from an earthed computer chassis being connected to often single ended floating DACs followed by earthed amplifiers.

Re:solder 3 wires and get good toslink out

[TheGratefulNet]

opto IS the universal spdif interface, that's one reason.

no ground loops. MAJOR MAJOR reason. can't emph this enough.

finally, any DECENT dac will reclock via its local fifo. that renders the opto jitter nearly irrelevant.

the latest chip I'm playing with (wolfson wm8805) is a nice reclocker. takes in spdif and puts out spdif, like a bridge, but lowers jitter due to buffering ON the chip.

with this guy in series, cable jitter should not matter. SHOULD not matter. I have not tested this but I do trust wolfson as they have 'real scientists' working there and I do trust the work they do. wolfson chips are some of the best in the world and this spdif receiver (to i2s or to spdif) has really low jitter timing specs on its output.

(ob disc: I am designing and building an spdif switch box that uses this wolfson and have put a good 6months into it so far and still plan to move ahead with the pcb design and manuf. it will also have arduino open-src code to control it and have an LCD on the front, IR user-programmability and up to 8 inputs of toslink and coax, maybe even xlr if I can fit it in.)

Re:solder 3 wires and get good toslink out

[mako1138]

I've had trouble sourcing TORX/TOTX parts lately. Can't find them in stock anywhere.

Re:solder 3 wires and get good toslink out

[thegarbz]

RCA IS also the universal interface. Take a look at the back of your TV, DVD player or even computer in my case. It's the RCA socket with the orange colour code.

The ground loops thing is a very real problem, but just like you say any DECENT dac will reclock the incoming data any decent dac will also have pulse transformer isolated inputs on the S/PDIF line. This is actually a technical requirement of any studio level equipment conforming to AES/EBU which is essentially a differential version of S/PDIF.

As for the reclocking, it's a bit of a misnomer. It doesn't eliminate jitter it just changes the form of it. The reason is that it needs to either run in asynchronous mode or drop samples in order to work. Asynchronous reclocking has a different but equally destructive sound signature to jitter, and funny enough the signature varies with incoming jitter. That's a fundamental problem for the audio industry. The bigger the buffer the less severe the impact but the more useless the DAC is as a real time audio device.

There is one DAC that I've seen eliminates jitter. It reclocks synchronously via a huge buffer and has a BIG warning that comes with it saying you can't use it for TV, computer or Radio as it has a 15 second buffer delay and relies on songs being roughly 4-5min long so it can flush this buffer without dropping samples.

There's no silver bullet, actually quite the opposite. There's active arguments in the audio community whether it's better to do ASRC, or to skip that altogether. Naturally no one has committed to a double blind test :-/

Re:solder 3 wires and get good toslink out

[greatgreygreengreasy]

Is this DiY guide still available? I've looked for cm102 parts, it looks like Newegg only has cm106 based devices, are these the same? Thanks for your help.

Re:Hypothesis Number Two: Is There a Cyber Stalker

[Anonymous Coward]

Do you have a newsletter?

Comment removed

[account_deleted]

Comment removed based on user account deletion

Let me be the first to say...

[Anonymous Coward]

*DUH*

Haven't we known this for years and years?

Re:Anyone know about laptop sound quality benchmar

[TheGratefulNet]

search on rmaa (rightmark audio).

a lot of us use it to measure our own hardware builds and designs. its quite good even though its not a $10k package that those that have deep wallets use.

you can do loopback tests of your analog system (line in to line out, short wire that you buy or build) and it will show your worst case. you can do digital loopback too, if you want. or digital out (to a dac) then analog in to a good a/d box, so you measure outboard dac performance.

like always, you need better gear to be able to measure lower end gear. the a/d and d/a matters, of course. but you don't need to spend a lot. m-audio firewire audiophile works well. the emu 0404usb box also is test-gear quality. those work with rmaa and get you going on-the-cheap.

Re:Avoid Asus AT ALL COSTS

[gman003]

How is this off-topic? The article, even the summary states that this is about Asus discrete audio cards. And I address not just general reliability concerns, but also audio-specific problems.

Honestly, sometimes I wonder if /. moderators are paid shills, or just really, really dumb.

Re:Given that Realtek et al can't make a good chip

[LeperPuppet]

They can't write decent software either.

I switched to a $35 Asus card and was pleasantly surprised. Better quality sound and no more flaky drivers.

I think the point is for people without stereos

[Sycraft-fu]

Many people use headphones straight on the system, or PC speakers. For that, a soundcard can be a cheap benefit. Is a receiver better? Sure, but then they are more expensive. You can always find better for more money. I'm quite partial to my 7.1 setup on my computer but I'm not going to suggest it to most people on account of the extreme cost.

So if you are a headphone type, a cheap soundcard can be a very worthwhile upgrade.

Also if S/PDIF is your thing you've no need for an external soundcard in most cases, generally the onboard ones have it these days. Since that bypasses any converters, there's no reason not to use it if that's all you want.

Further if you've a newer receiver that can do LPCM audio over HDMI most new systems can too. All the newer Intel, nVidia and AMD videocards can do that.

When you've a receiver, there are many choices. However for those that don't wish to spend the money, $50 on a soundcard (internal or external) is money well spent.

Not really impressive review

[gweihir]

These are not audio people and did not have an audio-expert look at their write-up. Why. They got the very well known very-low-cost / not-very-good audio OpAmp NE5532 P wrong as NE55329. No audio-expert would make that mistake. It is not a number, it is an identity that experts immediately recognize.

I have to say that this puts a big question-mark on the whole test for me.

Re:Not really impressive review

[Anonymous Coward]

NE5532 is pretty well respected as the cheapest decent headphone op-amp. Something 10% better than it will cost 300% more.
 
  link

Re:Not really impressive review

[jmak]

Also, according to their measurements, the conclusion should be more like "There's almost no point in spending $50 on a sound card, if you care only about playback, not recording."

Re:Not really impressive review

[MrL0G1C]

Their analysis of the RightMark audio benchmarks are also complete fail, they do not seem to either understand the Decibel scale / they can't read a graph / they don't understand the limits of human hearing or how Psychoacoustics fit in with the graph.

http://en.wikipedia.org/wiki/Psychoacoustics

For instance:

check out the Xonar DGX in the frequency response plot. Again, it starts dropping at lower frequencies than the competition

A less than 1/2 a Hz difference in a 10,000Hz frequency is not going to be audible, stupid people.

And:

Also worth noting: the higher noise and distortion exhibited by the integrated audio in several of the graphs.

The only way you'd actually hear any of the noise is if your amp+speakers where out-putting at ~150Db but not playing any sound (noise only), unlikely I think.

As for the distortion level, it's far too small to be humanly perceptible.

Re:Not really impressive review

[mwvdlee]

I noticed you used the wrong punctuation mark after "Why". It should have been either a question mark or a comma (in which case the next word should not have been capitalized).

I have to say this puts a big question-mark on your whole comment for me. ;)

Re:Not really impressive review

[gweihir]

Since the core of my posting was not about correct use of the English language, you can rest easy. I would definitely not call myself an expert in that area. I learned English in school. ;-)

Re:Not really impressive review

[gweihir]

Yes, you are entirely correct. I stopped reading after I saw the OpAmp naming fail. Re-reading the article I agree that their measurement "interpretation" is completely clueless. "Stereo Crosstalk" on Page 10 is also funny. The red line there is not even a plot and looks like the dots were corrected wrongly. Seems nobody actually understood what the point of these plots is and what they should look like.

There definitely was nobody with a clue about audio measurements involved. The whole article is bogus.

My personal observation is that the only thing in integrated audio that may be problematic (due to noise) is microphone input with mics that need the "microphone boost" (i.e. low-impedance ones that produce a low signal level). I have a preamp with a MAX410 OpAmp lying around whenever I need a bit better quality (rarely).

Re:Not really impressive review

[mwvdlee]

Yup, spelling is only a small part of a comment.
As is an Opamp only a small part of a sound card. Especially for a sound card in the $40-50 range.

Re:Not really impressive review

[gweihir]

Not at all. OpAmps are critical. The OpAmp mistake is about as serious as confusing a Porsche and Fiat. Nothing anybody halfway competent in the subject area would ever do.

You also seem to have completely missed that the "NE55329" is not a spelling mistake, but a misreading of the P, something nobody that understands audio electronics would ever do.

You do not get what this is about at all. Maybe if no clue, STFU?

Re:Not really impressive review

[mako1138]

Mod parent up, op-amp hype is rife in the audiophile community.

Re:Not really impressive review

[gweihir]

Audiophiles are typically not sound experts at all. The NE5532 is about the cheapest you can use without really damaging the signal.

Re:Avoid Asus AT ALL COSTS

[Anonymous Coward]

religious rants are offtopic in this discussion.

can that possibly matter with optical digital cabl

[ClioCJS]

I use optical digital cables to connect my PC to my stereo. Could the distance really even matter at that point? It's a pure digital signal. Why are the blind listening tests not done with pure digital signals?

Re:Given that Realtek et al can't make a good chip

[sethstorm]

I know that too well given the W520's use of it. The software stack in that can't even handle volume correctly compared to the T6x-era brethren that last used a decent sound chipset + stack.

Re:Avoid Asus AT ALL COSTS

[spauldo]

Just a guess, but I'd say it's because there's stories just like yours for every manufacturer out there*.

Case in point: hard drives. Ask 20 people what hard drives they've had trouble with, and you'll find they pretty much average out as all of the companies having issues. I use Seagate, but you'll find a lot of people here who swear they're the worst drives on the market.

Another example: T-Mobile. I had nothing but trouble with them. They would, for no reason, forgo the automatic draft from my Visa and then shut me off for non-payment. Their customer service was horrible. When I called them at the end of my contract and told them I wanted my service cut as soon as the contract was over, the sales drone threatened me that if I didn't pay the final bill, they'd sue me. I hadn't said anything about it up to then, so this was just out of the blue. (Of course, I knew he was full of shit, and intentionally didn't pay my final bill because of it.)

With all that, T-Mobile has an excellent reputation for customer service and very few people I know have issues with them. Go figure.

The article was about the difference between soundcards and integrated sound, and just happened to use Asus cards for the testing. Your last paragraph was on topic, but the rest of your post wasn't.

* There are a few manufacturers that have earned widespread derision, like PC Chips for its fake cache chips or SCO for judicial douchebaggery. Asus isn't anywhere near that level.

It's worth it if you listen to music

[DogDude]

If you listen to music... not compressed pop crap through itunes or something similar, but actual music, I've always found it worthwhile to get a good external sound card of some kind. Right now, I use Echo's Indigio IO that makes a LOT of difference going into my big Yamaha stereo receiver, and out through my pair of 4' JBL's. It's not an expensive setup, but a few hundred bucks at the PC, from my experience, eliminates the audio quality bottleneck that I've found on every standard PC/laptop with integrated audio.

No....

[Anonymous Coward]

I don't think so.

When every motherboard started comming with all kinds of audio. I was still dragging around fairly high end sound card hardware.
I thought it mattered. It does not.
For most games and normal use using headphones and small powered speakers. The onboard sound chips work fine.

A sound card is not worth the power it wastes in most cases.

Looks like some audio makers are trying to drum up business now that they're almost dead.
And as for creative labs... hahah. fuck you and your mess.

I used to use Creative soundcards...

[antdude]

... And they (original, 16, Live, Audigy 2ZS, etc.) were still better than the onboard audio IMO even with my poor hearing (wear an analog mono hearing aid). I also have a Logitech I used to game a lot so I wanted hardware EAX. I had to dump Audigy 2ZS because of the lack of old PCI slots on the newer motherboards/mobos., so I decided to try onboard RealTek audio. I also don't game these days. Well, onboard's quality and subwoofer's bass were less on my Logitech Z-2300 speakers (2.1 setup and analog).

I still have not found a good one to buy that was fully compatible for Linux and Windows after 1.5 years. I just listen to MP3s, watch TV/television, movies, videos, etc. Onboard is OK if audio quality is not important.

It varies by service centre, and country I'd say

[dbIII]

Where I am ASUS seem to have the quickest turnaround (1-3 days) but the service centre is local and I've dropped items there in person. That's only from about three items over a few years though.
With others it's been well over a week at least and about three to five months in one case for a video card (eVGA, but when the replacement finally came it was a much better card worth about $200 more). If the stuff isn't dealt with locally it might even be a week or two before it's even shipped out. I'd say if I lived in a different city I'd think Toshiba are wonderful instead of the slow as shit impression I have, and probably think that ASUS are crap

HDMI Sound works better.

[bejiitas_wrath]

I use a large screen with HDMI input and I can output sound over the HDMI cable as well, this removes a lot of the noise that the integrated sound suffers from on my machine.

Re:HDMI Sound works better.

[Anonymous Coward]

On my laptop (which uses Intel 4500MDH graphics for HDMI) I get significantly more noise with HDMI than I do with the already poor analog jacks that have noticeable static. Newer drivers reduced the frequency of the periodic clicking I'd get on HDMI, but significantly increased the frequency of the complete synchronisation loss where all I'd get is a complete loud hiss even if the audio out was supposedly set to mute. My deskop (which uses AMD graphics) does not have this issue, so while HDMI can sound better, it's fairly finicky.

Sound quality = bass

[Anonymous Coward]

At least to most people these days. That and the loudness wars have gotten so bad that I distrust any mainstream test on audio quality. I've got a Clip+ and a pair of Beyerdynamic cans and someday I'll get myself a decent DAC.

Re:Avoid Asus AT ALL COSTS

[robbie73]

Dude, I think you have static electricity (charge) issues or you are just unlucky. I have had several Asus boards before, none of them had issues. I have an Asus ultrabook, I am listening to music on it while typing this... PEBKAC = Problem Exists Between Keyboard And Chair

PSU and THX

[MrL0G1C]

I bought a nice 520W Corsair PSU has lasted well for years now, only audible if you stick your ear next to it.

With a ASRock Z77 Extreme4 - has THX certified sound out, which I was a little skeptical of but it does seem to eliminate motherboard interference noise

Previously when using a cheap mobo audio, a browser page being scrolled up and down would cause noise, maybe not noticeable with small speakers but very obvious when connected to an amp + large speakers.

Not a cheap mobo / you get what you pay for.

Dumped the Realtek for a Xonar

[DrHappyAngry]

Ever since I bought this motherboard, Asus P8P67 Deluxe I had crashes with pulseaudio. Almost always if I tried to play multiple sound sources, like not stopping a video or music with you tube playing or something or running smplayer and audacious at the same time. This was under both Ubuntu, and later when I jumped to Fedora. Decided to bite the bullet and get a Xonar DSX, and no more crashes. The one thing is to configure it as 7.1 card in in the pulse settings, even if you're only using a 5.1 speaker setup, otherwise your speaker arrangement will be off. The other caveat is that the card requires a 4 pin floppy power connector. It comes with a molex adapter, though. Apparently a 1x PCI-E slot supplies less power than a PCI slot, so it needs the extra power. Nothing that can't be worked around, so I'm happy with mine. Totally worth it. I had stockpiled a bunch of old SB live and audigy cards for a long while, but since I dual boot for games, win7 didn't play as nicely as linux with those old cards. Those old EMU10k1 based chip cards were so awesome under linux. Actually had fantastic ALSA drivers that could do hardware mixing with no sound server, but don't get me started on that. I could go on all day about how the real problem with sound under linux is crappy drivers and we shouldn't need a sound server. But I digest... The xonar has been a great upgrade. I disabled the realtek in the bios, and the card was detected by fedora without having muck about with loading modules.

HF noise matters

[dutchwhizzman]

Because the HF has harmonics that are in the audible spectrum as well. Not only that, but it "dampens" the amount of dB you have left for signal/noise ratio. You can try and use a low-pass filter to filter it out, but since those are analog, they will also filter out part of the audible spectrum and not filter out all the hf noise, just dampen it.

Re:HF noise matters

[Rockoon]

Also remember that there is no such thing as a brick walled perfectly flat filter that doesnt have an infinite delay, either analog or digital.

If you go for the brick wall, you get ripples in the pass-band..
If you avoid the ripples, your brick wall becomes a slow roll-off.

The real solution is not to play this game. If you dont have frequencies you dont want then you dont have to filter them.... but thats hard when you've got a PSU and other electronics throwing noise out all over the spectrum.

The PC best audio experience I have had so far over the decades has been with a Logitec USB headset. If there is a source of noise in either direction, it has escaped me... I simply cannot hear it (nor is any mic noise visible in audacity.)

Re:HF noise matters

[thegarbz]

This is semantics anyway. Most amplifiers have some form of power supply rejection. You never get the switching noise passed fully through the gain stages, and the actual switching signals are typically very small when you look at a SMPS waveform. If you have are affecting your possible SNR by more than a completely trivial amount you should redesign your amplifier or replace your powersupply.

Re:HF noise matters

[TheGratefulNet]

PSR (power rejection ratio) is spec'd at various freq's.

at lower ones (60 and 120hz) its usually good.

but NOT SO GOOD at mhz class noise.

that's what switchers put in and you almost cannot take it out once its in the audio stream.

audio systems that use switchers almost always do it for cost reasons, NEVER for audio cleanliness. also to save weight (cost).

given a choice, not one single audio designer would ever WANT a switcher anywhere near his gear. not even feet away from it, turned on, on some other item.

look at the spectrum if you ever get access to such gear. you'd be amazed. and then look at the PSRR rejection. its not enough! good starting point but not high enough. and psrr on discrete designs is even harder (if you avoid opamps).

Re:HF noise matters

[thegarbz]

PSRR is measured post gain. If a system has enough bandwidth to amplify the switcher signal it can be designed to minimise switcher signal (i.e. balanced Class A rejection of in phase noise on power rails), if it hasn't enough bandwidth to amplify it then it can't actually amplify it and is incorporated in PSRR anyway.

There's several companies which incorporate switchers into high end audio gear, these are also companies which actually know something about designing a variety of gear rather than just throwing a Class AB high power amp in and being done with it. B&O comes to mind immediately, so did a company with a similar pedigree of Mark Levinson (but not ML, I just can't remember which one off hand). Typically the same companies which do this also experiment with Class D amps, and they do a damn good job of it.

Also weight and cost become the overriding factor if you start designing systems close to 1kW. The fact is that it's not impossible to do, it just takes a bit more finesse than the typical head stuck in the 1980s era tube amplifier designer will commit.

Re:Given that Realtek et al can't make a good chip

[Mashiki]

They can't write decent software either.

They can write a driver? I thought all they did was throw some kid fresh out of highschool that passed their pascal test, and said here's some Jolt.

Re:Avoid Asus AT ALL COSTS

[Mashiki]

I use Seagate, but you'll find a lot of people here who swear they're the worst drives on the market.

I can remember when that was true. Back in '99, the shop I used to work for would order drives by the case from Ingram Micro. Still sealed from Seagate with the security seals in place. Every drive we'd put in, would be DOA out of the box. We switched to Maxtor, about a year later it started happening with Maxtor, where about 50% of the drives were DOA. I know makers can have serious batch run issues, but the problem with Seagate was beyond stupid.

It sure was nice when you had the option of more than a couple of manufacturers on the market.

Outputdev?

[theshibboleth]

OK, maybe this is a technicality but it seems to me that this really belongs in inputdev and not the tag I see. In this case the input-generating machine happens to be a computer and this input is travelling to the sound card. The sound card then sends it to some other device after some processing. At this point the sound card becomes an input device for whatever is receiving the information. It is an acceptor and a generator. And any device that is a generator mentioned here is more likely to be a generator + acceptor than solely an acceptor.

If you have a receiver and Linux

[egork]

this will give you the highest return on investment if your receiver is capable of 24/96 and 5.1. No hardware changes necessary.

sudo vi /etc/pulse/daemon.conf (this is Ubuntu)
and change following entries this way:

default-sample-format = s24le
default-sample-rate = 96000
default-sample-channels = 6

Re:If you have a receiver and Linux

[fa2k]

Also,

resample-method = speex-float-8

gives you a better resampler (10 is max. quality at the cost of CPU, default is 3. Don't know if it affects latency, but I'd imagine it would.). PA will only output a single sampling rate, so for example all your MP3s at 44.1kHz will be will be resampled.

There are some good arguments against 96 kHz. Specifically, it is impossible to hear anything above about 22 kHz, and 44 kHz of PCM is sufficient to encode that. If your equipment is not perfectly linear up to 96 kHz, any error in reproduction of ultrasound can cause audible distortion.

Re:If you have a receiver and Linux

[Lumpy]

remember marketing speak. 96000 across 6 channels. I am certain that is not per channel. so 16Khz sample rate per channel. and that is typical from what I see on soundcards.

Almost NONE of them actually do what is claimed.

Re:If you have a receiver and Linux

[TheGratefulNet]

I think I tried it once and it did, in fact, cause everything going into my DAC to show up at 96k. I kind of liked that ;)

what I have not tried is 'how good are the sw resamplers' on linux. for that, I just don't know.

the other question is: how much better are the non-realtime converters? if I have 176k and 192k stuff and I want to save it all on another ('play') disk as all 24/96k, is it worth doing that or is realtime THAT good, now, that you never have to do this?

(my sound system stops at 96k and won't take anything higher and so I need to either bulk-convert all my 'higher bitrate' stuff down OR find a good realtime resampler.)

Re:If you have a receiver and Linux

[TheGratefulNet]

in my world (I am pretty much only living in a 2.0 world, just left and right from discrete channels) 96k does really mean 96k, on each of the 2 channels. and 192k and 176k and 88 and 48 and 44 and 32 (that covers them all) really means that bitrate. measured on the word-clock line, on the i2s wires. I've done it. my 'spdif meter' does just this and shows the wire's bitrate, for spdif 2-channel.

all bets are off when its non-redbook. if its dolby or dts, its not open-source or free and even the chips are not easy to get in single qtys. the DIY world and audiophile world stays with 'stereo' (2.0) for that reason, and almost that reason alone.

and so anyway, if you set alsa or pulse up for 24/32 bit, you get that. in fact, you can get that even at 44 or 48k rates. you don't need to jump to 88 or 96 or higher to get 24bit audio. (fact: the payload always had 20 bits of 'user space' for audio and 4 bits of 'extra stuff for future use'. in the last bunch of years, they combine those to make 24bit payload and all modern receivers and dacs notice this and can decode 16/20/24 bit just fine. even at 44.1 and 48k. fyi.)

Re:If you have a receiver and Linux

[Anonymous Coward]

Unless Ubuntu is heavily behind the curve it should have PulseAudio 2.0 which can switch between sample rates on-the-fly. Basically after silence the first sample rate that is played back will be used to set hardware rate if possible. That way no resampling is done for the particular stream, if other sample rates are used for other streams and switching is not possible (something is currently playing) then those would get resampled. To control which sampling formats are used (probably so that some crazy low sample rate is not picked there's two possible sample rates are defined by default - 44.1 kHz and 48 kHz which effectively cover all important cases - CDs are 44.1 kHz by the great Sony wisdom and the non-retarded stuff such as most TV programmes and at least some DVDs are 48 kHz.
Since virtually everything is 16 bit signed integer unless it has gone through software filtering (which would likely fancy 32 bit signed float that generally even expensive sound adapters can't handle) by using s24 you are effectively mandating resampling. Instead just stick to the defaults (same quality as Audio-CD + dynamic sample rate) and you have decent chances to avoid most of resampling when using PulseAudio 2.0 or newer. Saves CPU cycles, power, decreases latency and avoids loss of quality - just do it (or rather do not touch daemon.conf)!

Re:If you have a receiver and Linux

[Anonymous Coward]

I have a 5.1 receiver and Linux - how do I get this signal into my receiver digitally? As far as I can tell, SPDIF is either two channel or AC3/DTS only. There's no way to get six channels across a SPDIF connection.

MP3 killed Hifi

[flyingfsck]

The ability to compress and stream music proved to be far more important than high fidelity sound. So a better sound device is kinda pointless.

Re:MP3 killed Hifi

[Anonymous Coward]

The ability to compress and stream music proved to be far more important than high fidelity sound.

Unless you're listening to classical stuff or electronica, 320kbps MP3's will sound identical to an uncompressed file. The research has been done, the conclusion unavoidable unless you're susceptable to "magical thinking".

So a better sound device is kinda pointless.

The biggest improvement in audio quality would be acoustic treatment of home listening environments (minumum room size around 1500 cu ft). Discussions about amplifiers, loudspeakers and frequency response are utter nonsense for anybody who has not first dealt with the acoustics of their listening room. Again, we're into the realms of "magical thinking" or of ignorance paired with marketing for those who refuse to accept the physics.

Re:MP3 killed Hifi

[TheGratefulNet]

there's still the original sound material and you can CHOOSE to listen using non-junky (non lossy) methods.

its great to have freedom. I have both 'paths'; the mp3 lossy path (usually in my car or on portables) and the lossless path with either cd or actual rips to flac or shorten. when I listen to those, they do deserve better playback equipment and that includes the soundcard, the interconnect (to some degree), the transmitter receiver pairs at each end, the encoder/decoders at each end and so on.

the thing is, the hardware is not expensive. and so 'doing it right' is very cheap and easy to do, why not do it? the higher end stuff (cd, dvd, bd) benefits and your mp3 will sound as good and not any worse.

if your pc is not too far from your avr, there's little reason not to run an opto cable from the back of your pc to your home stereo and switch to its opto-input for music and movie playback. you can run usb fairly far, if you have to, and even repeat it with powered repeaters. then run the final length of opto cable, 50' or less (if you can help it). you can get a decent amount of distance and still be able to use the 'remote' avr in the next room this way.

laptops make things easy, too. a laptop to a usb/opto dongle and then opto toslink cable to your avr. an old laptop can be used. extremely old. like, very extremely (10yrs is still enough to do usb2.0, I think?). if it can pump data out at usb2.0 speeds, an ancient laptop can be just fine as a sound source driving an avr's toslink input.

doing it the modern way: mini-itx fanless pc. in fact, those have good enough onboard toslink ports, just use those! they are bit perfect if the os keep's its mixer out of it (do-able on linux, choose the 'hw' interface in alsa). the mini-itx atom fanless or e350 amd fanless boxes are my choice for building a nice audio server for feeding a home AVR via spdif. they draw all of 13 watts, too (using an ssd and linux 'voyage' embedded).

Re:MP3 killed Hifi

[bipbop]

True, though the selection of "difficult" material doesn't fit neatly into "classical stuff or electronica" like you suggest. And of course, 320kbit MP3 usually works pretty well on both of those.

(It's also true that some extremely trained listeners can beat chance in distinguishing 320kbit mp3s from originals. But this is only barely true.)

More usefully, it's nice having uncompressed audio so you can do things to it without noticeable degradation, as many activities involve an encoding step, and MP3 isn't designed for tandem encoding.

asus u3 and u1

[gl4ss]

I got both of them.. the sound is good, there's optical spdif and they offer dolby live(live encode to 5.1 dd).

the u3 is like 30 bucks too, very cheap.

the drivers could offer more options though(it would be nice if they had adjustable low-pass for the 5.1's subwoofer channel for example).. that's the weakest link of both of these usb dongles. but it's still better than splicing the audio from the hdmi output, which offers pretty much no options at all(nvidia).

it kinda sucks that nvidia had super good options and dd-live on the nforce2 boards and then dropped it. if they had same kind of options on their hdmi output that would be super nice.

Features, not sound quality

[Anonymous Coward]

Interesting review. As a hobbyist musician, I never thought people would buy separate soundcards for the difference in audio quality, because noise issues aside, the difference is negligible, and all that talk about better soundstages or fleshier vocals is just audiophile stuff. And most of the world generally considers audiophiles a joke.

I thought people buy separate soundcards for extra features, more outputs, less latency, to avoid possible noise problems etc. The article seems to miss this point entirely. I also think its worth noting that the differences in the frequency response graphs and such exist at frequencies a human hear cannot actually hear (20khz etc).

Re:Features, not sound quality

[Lonewolf666]

Actually, some people do to get decent sound at all. Some integrated sound devices are just so bad that you need to replace them with something better. See my other post about my own experiences:
http://slashdot.org/comments.pl?sid=3041333&cid=40956281
Two out of three sucked hard enough that a replacement was necessary.

Asus Xonar DG

[Anonymous Coward]

I am suprised the Xonar DG wasn't mentioned it's a great budget soundcard that has a built in headphone amp that allows you to run headphones up to I believe 150ohm off hand. This goes for $30 normally but is frequently on sale for $20 with a rebate. It hit's well above it's budget class if you use headphones for listening and is what I bought when it was recommended to me on Head-Fi.

Re:Asus Xonar DG

[fa2k]

DG is the PCI version of DGX. I bought a DG, but the headphone amp is not supported on Linux, and there is no hardware volume control, but that's OK because you have 24 bit to play with. It was also branded heavily as a "gamer" card, but I suspect that's just to get the music fans to buy the more expensive cards. Anyway, it sort of worked for me, I got the ST as well. That's more like € 160, but it sounds much better on my 80 ohm headphones, and it's great on my stereo speakers too. I would like to get the Dolby Headphone effect on Linux, but the Headphone Spatialization effect in VLC is also pretty good, and more subtle.

So Linux users beware, the headphone amp on DG(X) is not supported on Linux

Re:Asus Xonar DG

[fa2k]

it sounds much better on my

Sorry for the overall poorly written post. I have to correct this: not "much" better, just a noticeable difference. If the DG's amp had worked, I probably wouldn't have heard a difference.

Re:Asus Xonar DG

[Anonymous Coward]

> I bought a DG, but the headphone amp is not supported on Linux

When was that? The DG support in Linux is pretty recent - looks like it is from 2.6.38.
Now, I have a STX and *everything* is supported on Linux, I can even change the impedance, the DAC filter, etc. out of the box. Pretty amazing.
I don't know what is the Dolby headphone effect, but I guess it's a patented software filter. Would the same effect be achieved by bs2b?

Good point

[pablo_max]

Very good point regarding the amplification stage. Audio amplification is really no different than RF amplification, just a much lower frequency. Typically, one would use a band pass filter or the like to order to keep unwanted signals out of the amplifier.
Of course this doesn't help when you have harmonics of other signals which happen to be in this band. Which is why things like proper grounding and shielding are so important.

Re:Good point

[theshowmecanuck]

Speaking of amplification, I looked at the article and couldn't see any tubes (valves). Where are the tubes?

Why is that surprising?

[pablo_max]

After all, a digital circuit required an oscillator to work. That oscillator when not properly grounded will couple to other signal paths. Basically, anything which generates a frequency can couple onto another circuit. Even if the primary frequency is not interesting to the components of that circuit, the harmonics could well be.

Re:Why is that surprising?

[darkHanzz]

It's somewhat surprising since even though the digital audio is sent bit-perfect over the S/PDIF cable, the audio quality can still be less than an obviously not-perfect analog cable. Apparantly you have good systems knowledge, a typical non-electrical engineer does not / should not have that, and for those, the "24bit/96khz digital can be worse than analog" can be suprising.

Headphones and opamps

[Anonymous Coward]

If you want real "blow your mind" quality from your audio, buy a very good pair of headphones (I've had Sennhesier 595's for a while and they're very good plus very comfortable, the latter of which is pretty important if you'll be wearing them a lot), and get a good quality DAC which also allows you to switch the opamps; also, ditch your mp3 collection for flac (lossless) files instead (it definitely makes a difference, especially compared low-bitrate mp3s full of artifacts).

I got the Xonar Essence Once recently (probably overpriced, but very good) and the quality is excellent albeit not as much bass as I'd like; a nice feature this has is the ability to switch opamps (I believe the PCI Essence ST can do this too), for the I/V converters, buffers and even the low pass; so I decided to learn a bit about different opamps and try out customizations.

The base NE5332P's opamps are nice but a bit low on bass, so I've had fun checking out LME49720NA's (amazing clarity from these opamps, better in that respect than anything else I've heard yet, but sadly lacking in bass; these are best for the buffers though), then OPA2111KP's (much improved bass, big improvement overall compared to NE5332P, but missing the amazing clarity of LME's), and I'm waiting now to get some AD797BR's off ebay, which have a good rep.

You can get metal-cap versions of all these opamps as well, which are extremely high quality but more expensive; once I find the opamp configuration I like the best, I'll probably buy metal-cap versions to get even better quality out of it.

So yea, if you listen to music a lot and want to bump up the quality, you can have a lot of fun with a customizable sound-card/DAC, and some good quality headphones; it might cost a fair few hundred, but what you buy will last you well past a decade, and being able to upgrade the opamps is very nice.

Here is a good resource for reading up on opamp modding (primarily, in seeing what opamps there are to try):
http://forums.guru3d.com/showthread.php?t=255682&page=47

SB Live Trick

[gallondr00nk]

I dumped my Sound Blaster Live! 5.1 for Realtek ALC888 and never looked back. I never heard such clarity until I went with this onboard chipset.

The SB Live uses two different DACs for front and rear channels, and the consensus is that the DAC for the rear channels sounds significantly better than the front. I had a Live 5.1 and this was the case for me too (though I ditched it for an M-Audio Revolution a while back - great sounding card, woeful drivers).

If anyone is curious, try the kX Project drivers on Windos (*nix you can swap it in alsa).

Oxgen free copper?

[Anonymous Coward]

I hope these guys are using oxygen free copper on the pcb. It is well known that Oxygen (evil, evil) corrodes the sound making is flaky and well, rusty.

Re:40kHz? Really?

[Ironhandx]

20 Khz is detectable by most people as a buzz, no matter what is transmitted. If the buzz is clear enough and follows the rest of the band transmitting its fine, if not then about 50%+ of people will find something wrong with the music. However since its not a frequency broadcasted in all songs most people register it as a problem with the SONG or style of music and just don't listen to that type of music. This is also why a lot of people didn't/don't like the sound of the new high-strung low-cost stereos and previously computer speakers.

Re:Avoid Asus AT ALL COSTS

[marsu_k]

And on the other hand, I've been very happy with my original Transformer, even happier now with Transformer Infinity. Aren't anecdotes wonderful?

Components returns rates

[MrL0G1C]

I find this page handy:

http://www.behardware.com/articles/843-1/components-returns-rates.html

The site gets these statistics from time to time.

MP3 is more than enough for Hifi

[egork]

Provided you do some sensible bitrate. Hifi is specified as 40Hz-16kHz on -6dB for amplifiers, for loudpeakers it is even worse. http://www.pinkfishmedia.net/forum/showthread.php?t=121725&page=33

Re:Given that Realtek et al can't make a good chip

[91degrees]

They don't want to make good chips. They want to make cheap chips. Most of us don't really care too much about audio quality except insofar as we want to be able to distinguish sounds.

The bar to "acceptable" is lower than you suggest.

Soundcard? how quaint.

[Lumpy]

Just give me the spdif out. Optical please.

run that to a real external amplifier and DAC. All done.

All I wanted was something that sounded....

[3seas]

...better than the on-board audio. I got an ASUS DG for about $32 and I got better than I thought I would. Just wish there was a way to set the default output to other than FP headphones (which cable connectors I either don't have or need an adapter - forgot what the deal is here to not have the front panel audio connected to it)

For the audio pros... be sure to spend your extra bucks on gold plated cables, while I use simple copper lamp zip cord. lol

Creative...

[SIGBUS]

If only Creative wouldn't have shit in their customers laps when Vista rolled around and actually offered proper drivers for the thing.

That sounds awfully familiar. I have an E-MU 0404 USB audio interface - the sound is absolutely wonderful, but they never got beyond a poorly functioning beta driver for Windows 7, so I'm stuck with XP, or getting a different interface such as a Focusrite. The ASUS cards seem like they'd be nice, but I don't want baked-in equalization that I can't change.

Recent Linux kernels seem to support the box, but no Linux audio player can manage my music library as well as Foobar2000, and WINE won't let me go to 24-bit 96 kHz.

It's a shame Creative took over E-MU.

Case Bloat

[Guppy06]

There's much to be gained from upgrading to a discrete sound card, though.

Yeah, like yet another occupied expansion slot.

With the availability of mini-ITX boards and their whopping one (1) expansion slot on the board, these sound boards had best give me eargasms before I'd be willing to commit to consuming that increasingly limited resource.

And the real kicker is that it's these small form factor boards that are used in home theater PCs that are hooked up to serious speaker systems to begin with. It's the full-sized ATX boards with the 10" fans and case lighting that typically gets hooked up to integrated monitor speakers, or worse.

Re:can that possibly matter with optical digital c

[TheGratefulNet]

distance generally won't matter MUCH on toslink cables.

yes, jitter matters but only at very high end. your avr system, even if its a $2k monster, is still never going to be 'good enough' to care about picosecond level jitter in the spdif stream.

toslink does blur the digital audio and it changes timing slightly, randomly. that's jitter.

shorter cables do less damage.

at least on plastic toslink, there's no concept of reflections. back-energy does not happen on toslink. but it DOES happen on coax/spdif. shorter does matter here. think of it this way: you send a signal to the far end of the coax (again, not opto, but coax) and it sends most of its energy there but reflects back some. that takes some time to travel along the cable back to the start. it then bounces back again, along with new energy from the last pulse of the transmitter. this goes back and forth and blurs the 'location' or timing of the 1's and 0's.

now, if your DAC system fully and completely locally (!) reclocks, you are fine. if not and if it DEPENDS on the timing of each and every 1 and 0, it would 'dump out' the 16bit audio word at the wrong time since one of those 'clock edges' was off by a bit, due to the reflection blur. it happens but its test-equip level, not 'wow, that sounds horrible' level. very subtle but once you have $10k-class spkrs (etc) you CAN hear blurring of the timing.

long answer: but in real world, you don't care about cable length in digital audio. best to stay 6' or 10' or less. if you go farther, you MAY want to consider a bridge (like data link bridge; fully receives spdif stream then recreates it via a receiver/transmitter combo; not a repeater but a full recieve, digest, regurg, retransmit pair of rx/tx chips).

well

[__aaqvdr516]

Going to a nursing home and asking 90+ year old battleship gunners to test audio quality will lead to dubious results.

I don't have superman hearing. I have a Creative Labs card with Medusa NX 5.1 headphones. If you take my sound card away and force me to use on board audio, I'm going to cry like a little girl. On board audio is missing things like: bass redirection, crossover frequency contols, bass boost.

If you can't hear the clipping that occurs at high volume using on board audio, I don't know what to tell you. It's terrible.

Digital out is a different story all together.

There're more important things than that.

[Anonymous Coward]

Poorly designed circuits will introduce enough noise and distortion that your 16-bit DAC might as well be a 4-bit DAC.

Also, while the grandparent mentioned distance from the ATX supply, it's really nearby digital logic that ends up in the audio. It does no good to have a 16-bit DAC if you can't supply it with a voltage with less than 0.00001 volts of noise, and that requires a lot of power filtering. You also need a circuit designer that realizes that the circuit traces themselves are part of the circuit. The traces have resistance, creating voltage dividers everywhere they go, thus in well-designed circuits you'll often see many traces run parallel to each other only to finally join at their destination. This is done so that the voltage drop created by the current draw of one circuit doesn't affect the voltage available to another circuit. There are a lot of little things like that which have to be considered if you want 16-bit precision out of your audio circuit. Any random 16-bit DAC will work just as well as any other. Its the circuits people toss them into which end up ruining the audio.

That said, unless you can hear your mouse in your audio, the worst part of your system is probably your speakers. Buy some nice headphones.

Re:Avoid Asus AT ALL COSTS

[bigstrat2003]

I disagree. I have purchased, in the past year, two motherboards, a video card, and a Transformer tablet from Asus. All have performed perfectly, with no issues whatsoever.

Just get a Mac

[Kagato]

I think Apple is the only computer maker out there that actually gives a crap about audio. My load the onboard audio from most PC makers is horrible. You also get TOS out on all modern Macs (The headphone plus is also a Micro TOS out.). Still, in this day and age you shouldn't have to buy an external sound card. The audio should ben engineered from the start not to be crap.

Switched mode PSUs..

[Anonymous Coward]

switched mode PSUs.. can supply decent clean power - if correctly filtered, and not made in China - LOL
the second main thing i hate about switched mode PSUs - (the first being RF noise) is the fact that,
well, they're always operating about 1 microsecond from total disaster .. extra crowbar / zener protection
is a jolly good idea.. IMHO.
Soundcards, are in a horrible environment, no real geting away from the fact, quality DAC, screening, high
losless sampling rate and high dynamic range are needed, before you start to make serious audio from a
'soundcard' irrespective of device, OS, media, format etc before the real audio gear. Its all a moot point though
if its being listened to by a cheap components, sadly most people dont care. sigh.
   

In other news:

[MossStan]

Upgrading your graphics card substantially improves the graphics on your computer. In addition, upgrading the memory on your computer directly improves the memory on your computer. Oh and the processor...

DAC Sound Card

[Sirusjr]

I don't know why anyone would use a cheap sound card when you can get the same functionality from a quality DAC. Just plug it into the USB and plug your headphones into that and you are set. If you want to output to something else, just use the Coaxial or Optical outputs and keep everything digital. My NuForce uDAC has been my primary PC audio interface for years and it works beautifully.

Re:Given that Realtek et al can't make a good chip

[Anonymous Coward]

They can't write decent software either.

They can write a driver? I thought all they did was throw some kid fresh out of highschool that passed their pascal test, and said here's some Jolt.

Hey, leave off the Pascal!, the majority of the crap code I see nowadays comes from 'kids fresh out of highschool' who think they're C++/C# coders..or worse, VB coders, and that they can do anything..I didn't know anywhere still taught Pascal, thanks to all the crap thrown at it over the years by the C/C++ mob.
(For the record: started programming in assembly, then BASIC, then FORTRAN and ALGOL, then Pascal, C etc..)

A story about drivers, and I'll have to be careful as I've signed a NDA here. We identified an issue with a certain company's drivers for a certain product, so we contacted their Eu office, started to explain to one of the engineers the problem we thought might exist, the engineer stopped us before we got to the point and then told us exactly the issues we were having, and yes, it was indeed down to a driver bug, and they've known about it for some time. He then told us that they'd told the main office about this, including a trivial fix to the driver code they'd come up with, but the company wasn't going to update the driver to fix the issue as, for the target market, the device drivers functioned 'adequately' (their word, not mine). Needless to say, as this was a rather critical bug as far as we were concerned, we went elsewhere..

Re:DAC Sound Card

[gl4ss]

quality dac is just a fancy name for a soundcard in a box(or rather quality dac is just fancy speak for quality soundcard, soundcard usually being a dac, except nowadays you might not want to use the dac on it and just have it output digital straight to your amp).

asus makes couple of those too.

Another Nyquist Neophyte.

[Anonymous Coward]

It really is a good example of "A little knowledge is a dangerous thing" here.

That Nyquist limit is set to 44kHz to get 22kHz audible tone, but that audible tone is not faithfully reproduced in anything other than frequency. And then only if you can consider them as a single pure tone.

Timbre, shape (i.e. mixed frequencies as inherent in the shape) and volume all need higher sampling to get right. Why do you think a tin whistle at third C sounds different from a flute at the same note? How do you work out fortissimo and pianisimo without knowing the volume accurately?

Therefore sampling at 100kHz allows more accuracy in things other than frequency and those are awfully important to music. It's not just pure sine waves you know.

Maybe due to the amplification method

[Anonymous Coward]

I can't remember which way round it is, but digital op-amps amplify even harmonics and valve analogue amplifiers amplify odd harmonics. Since you hear mostly odd harmonics in nature, we hear odd harmonics as more natural and clearer. You can get over the problem with using even harmonics by producing more of them (you then need double the frequency and you effectively mute half of them) but this requires more sampling rates and more power for the same result.

It's why valve amplifiers sound REALLY loud at 15-20W RMS whilst you'd need a 70-100W digital amp for the same apparent sound.

Great PC Sound Needed?

[HArchH]

Let's just concede that the separate sound cards make better sound than the integrated sound chips. The question to me is, is that improved sound quality important? For me, my PC is not my home entertainment center's audio system. My PC is not my portable audio system. Skype-like video conferencing has very low quality sound, and output text to speech audio is barely understandable, anyway. I don't take the output audio wave forms of my PC and run them into audio signal quality meters though I might take the wave forms and run them through audacity to examine them. I play the PC audio through low-cost desktop speakers from Creative which are not, by a long shot, "hi-fi speakers". When they break, I throw them away and get replacements. I'd never consider replacing the foams on those speakers, like I do my real speakers. Why would I consider spending $50 for improved audio wave forms?

But why?

[Anonymous Coward]

I'd suggest saving the $50 on a "better" audio card and put that money in an incrementally better amplifier. There are plenty of good quality amps that accept digital in and have onboard multi-channel decoding. Keeping the audio output digital to the amp means there is less points of opportunity to degrade S/N. Current onboard audio solutions have for many years been more than adequate when just using their all-digital path.

I had a Sounblaster Platinum Pro card...

[Anonymous Coward]

...giving me MIDI IN/OUT and other features. I have not seen it again. Anywhere. Have not seen anything alike either. Myvery first soundcard was... a sound CARD. Integrated audio feels cheap compared to filling up specialized audio needs when you need it. Now I need something similar and even more advanced than my Soundblaster for a couple of laptops but there is no place to find it! Go to the very big electronic stores and you ll get all you can get. So the question seems out of place.

Re:Hypothesis Number Two: Is There a Cyber Stalker

[Anonymous Coward]

Do you know HiLJ?