1. Please take 30 seconds to register your free account to remove most ads, post topics, make friends, earn reward points at our store, and more!  

Definition of External Sound Card

Discussion in 'Recording Gear and Equipment [BG]' started by JonahTheAmazing, Jun 26, 2012.


  1. JonahTheAmazing

    JonahTheAmazing

    Dec 19, 2010
    I have been looking into home recording and have come across something I don't understand. What is the definition of an external sound card, and what makes it different than an internal sound card? I ask because I have heard several people refer to USB audio interfaces as "sound cards". When I type "external sound card" into Sweetwater's search, it takes me straight to the audio interfaces page. It seems to me that sound cards and interfaces are two very different products. I come to TB for answers.
     
  2. BKBassDude

    BKBassDude

    Jun 17, 2009
    Brooklyn, NY
    I have heard interfaces referred to as sound cards as well. I believe it is because it essentially disables or replaces the internal card when it's plugged in. All sound goes through the box. There are people who don't record who use usb sound cards to enhance the quality of audio their computers can reproduce.

    But yes, an interface obviously gives you much more functionality than a traditional "sound card".
     
  3. Freight Train

    Freight Train Earth-based Alternative Scientist, Sex Researcher Supporting Member

    Feb 25, 2012
    Dallas, Texas
    The deal is, a sound card in PC terminology is a card in you PC that is a digital-to-analog convertor (DAC) for sure, which gives you an analog line and headphone output so you can play your WMP or iTunes or whatever. It's also possibly an analog-to-digital (ADC) convertor as well, allowing you to digitally record from a mic or a line-level source like a bass preamp output etc. It possibly also has digital in's and out's allowing you to record to & from digital devices. If it has both digital-to-analog as well as analog-to-digital capabilities, it's called an ADAC.
    So when someone is taking about an 'external sound card', what they are talking about is a stand-alone box that can perform some or all of these functions. It will have a USB and/or Firewire, and soon if not now probably HDMI connector (if you're up-to-date and on board with Mac) to interface with your computer. You will normally get a much better sound quality in an external DAC or ADAC than in a sound card, because sound cards are heavily engineered for the gaming industry, so you're paying for all manner of surround-sound and fx functions that you don't need at all if what you're looking for is high-quality audio playback or recording.
    Here's an example of a very high quality USB DAC, which some may call an external sound card, but not anyone involved in professional audio production or audiophile hi-fi. Calling it an 'external sound card' is a typical lame last-gasp attempt from PC land trying to keep their antiquated technology somehow relevant. A DAC like this is the same quality as you'll find used in mastering rooms. http://www.guitarcenter.com/Benchmark-DAC1-USB-D-A-Converter-485677-i1531365.gc
    Here's an ADC from Benchmark, allowing you to record in stereo. http://www.musiciansfriend.com/pro-audio/benchmark-adc1-usb-a-d-converter
    Here are some ADAC's, which can playback as well as record. http://www.proaudio.com/index.php?cPath=130_222

    Not all DAC/ADC/ADAC's are this expensive. If you're just interested in stereo playback and recording, there are some excellent ones in hi-fi land for the $150 - $200 range that will outperform internal PC sound cards in the $500 range by leaps and bounds
     
  4. seamonkey

    seamonkey

    Aug 6, 2004
    You really don't need more than CD quality sound.

    Sound cards built on MotherBoards these days easily do HD quality audio. What they lack is level/gain matching to bass/guitars, and to normal mics and they have the little 1/8" jacks. They work better than most people would have you believe. Just use a small mixer to fix the levels. They all seems to support WDM/KS drivers which can be converted to ASIO with ASIO4ALL if your DAW software needs it.

    Audio interfaces, like on USB 2 is fine for up to 8 channels of CD quality audio. With pro connectors, and accept all kind of levels and mics. In most all cases they have their own supplied ASIO drivers, and also support WDM/KS. I mention this because with WDM/KS you can mix and match all the inputs and outputs in your DAW, including the ones on the motherboard. ASIO usually restricts you to one interface at a time.

    Firewire is also an option if you have it.

    If you have a desktop, a PCI/PCI-e sound card also called interfaces are by far is the ultimate sound card. For a low price. But they tend to have fan-out cables that are cumbersome. You can put together a high quality recording computer all for under $400 using a desktop.
     
  5. JonahTheAmazing

    JonahTheAmazing

    Dec 19, 2010
    Thanks for clearing that up. Talkbass knowledge! I hope others find this useful too.
     
  6. Others have given good info. I disagree with only one word in the quote above. "disables" IME most, the great majority, of audio interfaces... USB, firewire, etc.. do not automatically disable the internal sound card in a computer. The disabling usually needs to be done by changing some setting in the computer. Manually. It doesn't happen just by plugging the thing into the USB port.
     
  7. I think a true "sound card" can produce sounds via synthesis chips.

    Answer this, can you plug in a keyboard controller to an audio interface and produce an FM synth tone? I suppose you couldn't because it doesn't have the synthesis chips to produce it's own sound.

    An external sound card (audio interface) is a misnoner because it cannot produce sound on it's own. This is a marketing thing to get older users to buy their stuff. It's like solid state amps that claim to have "tube tone""TransTube""Vintage sound" ect...

    An audio interface is a device which lets you connect an external instrument/mic to it so that the PC can process the sound digitally. An internal sound card needs no external instrument, not even a keyboard controller, to produce sounds and music.
     
  8. Freight Train

    Freight Train Earth-based Alternative Scientist, Sex Researcher Supporting Member

    Feb 25, 2012
    Dallas, Texas
    Don't quite understand that statement. If seamonkey is saying you don't need anything beyond a 44.1k/16 bit ADAC, then I have to disagree, unless you have no interest in progressing to HD audio. If you are just recording to get ideas down and songs written, or just as a hobby, then that'll work. But pro audio production moved beyond 44.1/16 10 years ago or more. If you also use your computer for your hi-fi music player as most do, there is more and more music available online in HD, meaning at least 48k/24 bit, but preferably 88.2 or 96k/24 bit, and even 176.4 and 192k/24 bit (which is useless in my opinion). 96k/24 bit has pretty much been established as the standard for consumer HD audio. So unless you have a 'sound card' or external DAC that can play that, you'll be missing out. And there is definitely a difference that anyone should be able to hear, given a good enough playback system, and barring hearing damage.
    Most all qualified mastering rooms are now urging clients to produce their music and submit their mixes in 96/24 resolution, even if the project is solely for CD or mp3/AAC download at the moment. The project will be mastered at 96/24 and then downsampled/bit reduced to 44.1/16 for CD and data compressed formats. This allows the artist the option of offering their music in HD at a future time if we all come to our senses and HD downloads do indeed become more common. Regardless, a project that has been properly mastered at 96/24 will still sound better after downsampling to 44.1/16 for CD than if it were produced at 44.1k from the start. And it is always advisable to have your masters archived in absolutely the highest resolution possible.
     
  9. Freight Train

    Freight Train Earth-based Alternative Scientist, Sex Researcher Supporting Member

    Feb 25, 2012
    Dallas, Texas
    Yes, there's some confusion in terminology there. The function of an internal sound card most people use is to get analog out of the computer for playback, and analog into the computer for recording. But as Darius is saying another function of them can be for synthesis. That's another reason, as I said, that if what you are looking for is high-quality audio recording and playback, your best bang-for-the-buck is in an external DAC or ADAC, because with the sound card you're paying for a lot of things, like the synthesis Darius brought up, and the surround sound/fx capabilities for gaming that I talked about, that you may not need. That's why in terms of sound quality you can normally spend less for an external interface and get much better souind quality.
     
  10. mrbell321

    mrbell321

    Mar 26, 2012
    N. Colorado
    This makes me wonder... has anyone pulled apart or examined one of the high-end "interfaces"? I'm curious what chips they run on and if the important hardware in there is so different from a generic, onboard soundcard.
     
  11. seamonkey

    seamonkey

    Aug 6, 2004
    I use to think so myself, but in actual practice 44.1/16 is all you need. There is no loss at even this rate. It was blind tested decades ago when CD standard was established. 48khz is just a gimmick to fool people into thinking HD is improved in audio.

    It makes good selling to point for interfaces to have super high sample rates, but it does nothing for sound. Disk space these days is cheap, but if you want to do 8 or 16 channels on a normal PC, you don't need it higher sample rates and bit depth.

    Here's an interesting paper. It actually hits on the super high sample rates can cause other artifacts.
    http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf
    And these guys sell high end AD/DA converters.

    This is a good watch,


    The L-R signal is actuall AM centered at 19khz, and so 15khz is the high frequency cut off point. Many are quite happy with how it sounds. Satellite radio can sell higher response, but we're only human.
     
  12. DuraMorte

    DuraMorte

    Mar 3, 2011
    I would completely agree, except that during tracking, it's always a safer bet to use 44.1/24. Keeps you from running your inputs too hot.

    And during mp3 encoding, use 24 bits then too.

    But for just about everything else, 44.1/16 is pretty much all you need. 48k is more commonly associated with video (not sure why), and therefore really isn't a gimmick per se.
    192k is, however, completely pointless for music.
     
  13. Freight Train

    Freight Train Earth-based Alternative Scientist, Sex Researcher Supporting Member

    Feb 25, 2012
    Dallas, Texas
    I gotta respectfully say I'd do further research. I have been a mastering engineer for 20 years now. I hear the difference between 44.1 and 96k daily, at least 5 days a week. Probably 60% of the projects I do are submitted at 96/24 now. The difference in the detail in the 88.2 or 96k projects vs. 48 or 44.1k is unmistakeable, and if the 44.1k projects were mixed at 16 bit resolution, it's all the worse. If a project is delivered at 96/24, it's mastered at 96/24, then sample-rate converted to 44.1/24. Even though I use Weiss Saracon, which is the best SRC made, the loss is easy to hear. Those files are assembled, bit reduced and dithered to 16 bit for the production master DDPi for the glass master to be cut from. The loss from 44.1/24 to 44.1/16 is also audible.
    From there if the client needs mp3's or AAC's, I have specific software that helps identify the damage done at that stage, depending on what bit rate they require. Since the 44.1k SRC'd files are 24 bit, that allows me to go back and do minor eq tweaks etc. to compensate for those losses. Because if I further processed the 44.1/16 CD files, the loss of bits would cause a dramatic loss in quality, despite what whoever you're quoting is saying. This software allows for an immediate A/B between the 44.1k/16 dithered output of the mastering system and the resultant mp3 or AAC. There you can hear the biggest loss in all the stages I described, and none of my clients have failed to hear it.
    I'm speaking from working in the industry, over the last two years the effort has redoubled to urge iTunes and others to offer options for downloading at least 44.1/16, and hopefully 96/24 audio as well. Since the 4th qtr last year Apple has been meeting with major artists over this, and as of a couple months ago, Apple is requesting that all their music partners submit their masters at 96/24. Doesn't mean anything is gong to happen soon, but they are preparing for it.
    So I don't want to be argumentative, I'll just say if you want to apply the old "the average music listener can't hear it anyway" axiom, then to some degree what you're saying is absolutely true. But where that axiom is flawed is in the fact that, barring hearing damage, the average listener certainly can get into a better playback system, train their ear and eventually be able to hear it. The ear is just like a muscle; the more you use it, the better it gets. The more it's exposed to detailed, quality sound, the more it appreciates it and the more discerning and demanding it gets. What may seem like insignificant, or even inaudible nuances at first, will soon become easily distinguishable nuances and laster major factors in our appreciation of a particular recording. And I promise you and everyone those nuances are there in HD audio.
    Art, especially music, should not be dumbed-down in any way in this regard. You should always strive to produce absolutely the best quality possible, even though only a small part of the population may appreciate it. In the industry that is considered an obligation, because to future generations, the quality we produce is a reflection on our current culture and its art. When major labels were owned by people who knew and worked in music, nothing but the best quality possible was acceptable. Standards have lowered incredibly over the last 3 decades. That doesn't mean that every musician and producer shouldn't strive to meet those highest standards for themselves and the posterity of our music.
     
  14. seamonkey

    seamonkey

    Aug 6, 2004
    You are absolutely right!
     
  15. seamonkey

    seamonkey

    Aug 6, 2004
    There's not much to discuss with people who have Golden ears. You might be a multimillionaire from all your mixing experience. More power to you.

    For the not so millionaire, it's all there at 44.1/16 or 24, the research shows it, and so does experience. There is no compromise for people with good hearing which the standard is based on.

    It hurts nothing if you want to go higher, but most home studios are going to do fine and not tax the CPU by using a lower sampling rate. they'll get more channels. More channels would be more advantages than lesser higher sample rate channels. It's better to mix when you have multiple tracks to mix.

    Now for music web sites. I would like to see some music sites that would allow you to download multi-tracks of songs. So people could do their own mixing. They could include the automation track so you also get the producers mix. This will never happen but I can dream.
     
  16. Freight Train

    Freight Train Earth-based Alternative Scientist, Sex Researcher Supporting Member

    Feb 25, 2012
    Dallas, Texas
    Don't know what your agenda is, but you're not doing anyone any favors spreading that kind of nonsense. If you're happy with 44.1/16 or mp3's, then more power to you. But you should not discourage anyone who may want to improve themselves above that, as there's a lot of impressionable people on forums like this. As I said from the first, if you actually read what I said, 44.1/16 is find for someone wanting to screw around with garageband for the purpose of getting down ideas, writing, recording demos, whatever. But that is completely unacceptable for any professional production.
    Having a trained ear has nothing to do with having so-called 'golden ears' or being rich. The only thing you have to do to improve your ears, barring hearing damage as I said, is to expose them to well recorded music played back on an accurate, quality system for hours and they will eventually learn. I experienced this myself. It took two months for me to begin being able to discern the details the master engineer I assisted was pointing out to me. Anyone can get into a high-definition quality music system in small steps and rather inexpensively. Simple iTunes is a bit-accurate digital player up to 96/24 if it's set right. If you already have a computer and iTunes, you can get into a high-quality DAC for around $200, and near-reference quality phones for $99. Does that sound like something unobtainable except for us rich audio engineers? And if you think audio engineers get rich, that's proof you have nothing to do with professional music production. I don't doubt you for a second when you say you can't hear the difference between 44.1/16 and higher rates, or mp3's. I totally believe you. But you should not impose your limitations on others.
     
  17. seamonkey

    seamonkey

    Aug 6, 2004
    You win. Iggy
     
  18. lets get one thing str8 here-
    IT IS OF OUTMOST IMPORTANCE to have the highest possible sampling rate,because in the mixdown the transition to a smaller sampling rate will be less noticable and more natural/accurate(and this goes if,and only if you own good quality A/D/D/A converters)
    now,i agree,unless you are frikkin' R.Rubin or similar ace,you are not going to hear the difference as much,plus,i'm guessing you are NOT using the most expencive soundcard/DSP/A/D/A converters....so it's even harder to tell the difference between sampling rates,but get this-
    expencive A/D/D/A converters have ARMY GRADE parts,and you would really have to listen to it on a high end piece of monitoring to catch the subtle(but noticeable) changes in quality....
    it's always better to add depth and wideness-i.e working in a higher sampling rate enviroment(i'm using up to 48b/96khz)-so the transition,once you get to mixdown of a project,is less painful and more accurate.
    now,with cheap and mid/class soundcards,the trick is-it's just NOT worth it....either the difference is almost unoticeable,either you get glitches,pops,cracks etc when going on a higher sampling rate,bottomline-there's always ONE more problem to solve....but hey,that's why UAD costs that much...and that's why you have hordes of angry M-AUDIO fasttrack users,and others ofc.
    now,if you are going to listen to music on a serious monitoring system,you will MOST DEF. hear the difference between 16/44 and 24/96...trust me
    but if you listen to music on a genious 2.1 or similar......do I really have to go there?
    here's the trick with music production-
    ALL LINKS IN THE CHAIN MUST BE GOOD if not THE BEST POSSIBLE
    meaning-
    excellent soundcard,top notch PC,crappy monitoring system- no,no
    cheap soundcard,excellent PC,the best possible speakers- no,no
    everything top-notch except the speakers- big no,no
    so,try to figure what you need to have,considering your style/likings,and there you go...
    here's my chain at home-
    acer aspire intel dual core
    presonus audiobox usb
    korg d888
    kurzweil ks40a speakers
    i dont think this is the best possible,ofc....but it's
    cheap,and userfriendly....and does wonders...this presonus baby is great for the buck...
    ofc. in the studio it's a whole lot of a different story,but this IS about home recording....cheers...
     
  19. though it's logical,once you think about it-
    24b means the pc is taking 24 samples of an audio file per second.
    48b means it;s taking 48 per second
    and so on
    that leads to conclusion-IF you want a more accurate recording of your line-you WILL go higher....that's it
    now with KHZ,it's a bit odd....hence the cymbals go maximum to a 20khz,most people wonder-why on earth would i go higher,if there's nothing to record up there?
    well, 't is a bit tricky,and it would take a bit longer to explain,but let's just say it's safer to have the width like that....
    say your car is 6 feet wide
    would you drive it along the road that is 6 feet wide,or along the road that is 20 feet wide(where you have more breathin' space)??
     
  20. Stealth

    Stealth

    Feb 5, 2008
    Zagreb, Croatia
    You're mostly right except for two niggly details.

    The 24b you mention isn't the sample rate, it's the bitrate (24 bits per sample) that describes the minimum amplitude "step" that can be discerned. The smaller the number, the more granular the sound (the more distorted the waveform).
    On the other hand, the sample rate (represented in kHz) is what determines the amount of aliasing distortion (provided the ADC's input anti-alias filter doesn't have a steep slope it should have) and it affects the behavior of the filters and analyzers you use once the signal's digital.
    A high sample frequency (for each quadruple over the standard minimum 44.1 kHz) can provide the same effect as having a higher bitrate. If you have a 44.1kHz/16b signal and you rerecord it at 176,2kHz/16b, it'll actually be as accurate as a 44.1kHz/17b signal, giving you less granularity, a lower noise floor, and you can avoid the effect of the abovementioned "bad" anti-alias filter.
     

Share This Page

  1. This site uses cookies to help personalise content, tailor your experience and to keep you logged in if you register.
    By continuing to use this site, you are consenting to our use of cookies.