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Hardware

$50 Sound Cards Impress Versus Integrated Audio 245

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."
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$50 Sound Cards Impress Versus Integrated Audio

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  • by itamblyn ( 867415 ) on Friday August 10, 2012 @10:32PM (#40954505) Homepage
    Most of the improvement is likely due to increased distance between the amplification circuits and the noisy AC/DC power supply.
    • That, and larger circuits/ higher power means noise impacts less.
    • by ChrisMP1 ( 1130781 ) on Friday August 10, 2012 @10:37PM (#40954541)
      A decent ATX power supply really isn't bad. Distance from the noisy digital circuitry on the motherboard is important, though.
      • by TheGratefulNet ( 143330 ) on Friday August 10, 2012 @10:59PM (#40954649)

        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.

        • by ChrisMP1 ( 1130781 ) on Friday August 10, 2012 @11:20PM (#40954747)

          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.

          • by TheGratefulNet ( 143330 ) on Friday August 10, 2012 @11:32PM (#40954801)

            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.

            • by ChrisMP1 ( 1130781 ) on Friday August 10, 2012 @11:59PM (#40954871)

              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?

              • 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.
                • 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
                • 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.

                  • 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

              • 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 hap

            • 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.

          • 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 option

    • by TheGratefulNet ( 143330 ) on Friday August 10, 2012 @10:39PM (#40954559)

      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.

    • by Smauler ( 915644 ) on Friday August 10, 2012 @11:15PM (#40954733)

      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.

    • 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?
      • by TheGratefulNet ( 143330 ) on Saturday August 11, 2012 @09:21AM (#40956897)

        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).

    • by FunkDup ( 995643 )

      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.

  • by zonker ( 1158 ) on Friday August 10, 2012 @10:33PM (#40954513) Homepage Journal

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

    • by MrL0G1C ( 867445 ) on Saturday August 11, 2012 @03:17AM (#40955417) Journal

      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?

      • by gazbo ( 517111 ) on Saturday August 11, 2012 @03:43AM (#40955487)
        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.

        • by makomk ( 752139 ) on Saturday August 11, 2012 @04:14AM (#40955577) Journal

          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.

          • by gazbo ( 517111 )
            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.

        • by gazbo ( 517111 )
          "didgital"? WTF am I typing?
    • And when using just digital output, there is no difference at all anymore.
  • by TheGratefulNet ( 143330 ) on Friday August 10, 2012 @10:36PM (#40954539)

    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).

    • 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.

      • 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!

      • by epyT-R ( 613989 )

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

        • by TheGratefulNet ( 143330 ) on Friday August 10, 2012 @11:24PM (#40954775)

          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!)

          • by epyT-R ( 613989 )

            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.

          • by adolf ( 21054 )

            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 yo

            • 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.
              • by adolf ( 21054 )

                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 invol

    • 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,

    • 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.

      • 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 nois

  • by slackware 3.6 ( 2524328 ) on Friday August 10, 2012 @10:38PM (#40954545)
    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.
    • Except that I can still hear my mouse through every on-board sound system I've used recently.
      • 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.

    • by Smauler ( 915644 ) on Friday August 10, 2012 @11:22PM (#40954763)

      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.

      • by Mashiki ( 184564 )

        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.

        • by antdude ( 79039 )

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

          • by Mashiki ( 184564 )

            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.

      • 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.

    • by Mashiki ( 184564 )

      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 someti

      • 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).
      • by Luckyo ( 1726890 )

        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-5

    • Re: (Score:3, Insightful)

      by TaoPhoenix ( 980487 )

      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.
      Spen

      • by lowlymarine ( 1172723 ) on Saturday August 11, 2012 @01:15AM (#40955113)
        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.
      • 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.

  • by Anonymous Coward on Friday August 10, 2012 @10:38PM (#40954547)
    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...
    • 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).

  • 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.

    • 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 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.

      • by Mashiki ( 184564 )

        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.

  • $50? Try a $250. (Score:4, Interesting)

    by X0563511 ( 793323 ) on Friday August 10, 2012 @10:43PM (#40954577) Homepage Journal

    I have this for studio purposes [m-audio.com], 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.)

  • by TheGratefulNet ( 143330 ) on Friday August 10, 2012 @10:54PM (#40954629)

    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 [flickr.com]

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

    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

  • Comment removed based on user account deletion
    • 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

  • by gweihir ( 88907 ) on Friday August 10, 2012 @11:57PM (#40954865)

    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.

    • by jmak ( 409787 )

      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."

    • by MrL0G1C ( 867445 ) on Saturday August 11, 2012 @06:04AM (#40955923) Journal

      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 [wikipedia.org]

      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.

  • ... 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 stil

  • by MrL0G1C ( 867445 ) on Saturday August 11, 2012 @02:46AM (#40955341) Journal

    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.

  • 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 generat
  • 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

    • by fa2k ( 881632 )

      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, an

  • by flyingfsck ( 986395 ) on Saturday August 11, 2012 @04:13AM (#40955571)
    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.

All the simple programs have been written.

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