Effect of sound card quality on ripping m

I am trying to capture audio played over my sound card and rip right to mp3 with a program called Tunebite. I am wondering what effect, if any, my sound card has on the quality of the resulting mp3 file? Somebody told me this is all done in software and it does not matter that I only have the basic Sound Blaster PCI. thanks.

You must really like that program!
Actually, with the new method, you can make MP3's without a sound card! It bypasses the whole analog/digital conversion step, and it will make MP3's sound BETTER then the old method, even if you had the best card on the market. This is because it reads the sound data exactly as it is on the CD. Oh yea - it's probably 3 times faster too! (Depending on many factors... Can be 0X faster!) You don't have to wait for the songs to play!
Ya don't know what yer missin!
PD

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    Hi,
    What has been going wrong with your post ? You can use external USB sound card for your laptop. Here are few of them:
      http://www.shopbot.com.au/external-sound-card-usb/​price/australia/582295
    Regards.
    BH
    **Click the KUDOS thumb up on the left to say 'Thanks'**
    Make it easier for other people to find solutions by marking a Reply 'Accept as Solution' if it solves your problem.

  • Humming noise when I connect my sound card to my band mixer H

    Hello,
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    [size="2" face="Arial">

    phreemotion wrote:
    my tv takes rca...basically my question is....the RCA wires comign from my tv, how do i connect them to my sound card's flexijack...if thats even where it goesMessage Edited by phreemotion on 06-7-2008 04:45 AM
    You don't mention if you want to connect the card into TV through analog or digital path. I suppose you mean digital.
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    OR an adapter cable (RCA end by the RCA cable you have there (male/female)):
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    For digital connection, connect the cable through FlexiJack ... when you use stereo cable/adapter then the black/white cable carries the digital signal in the other end.
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  • New sound card (E-MU 1212M) won't record higher than 48kHz

    I'm running Windows XP Pro/SP2 with a 2.8GHz Pentium 4 processor (motherboard is Elitegroup 848P-A). I have 1GB of RAM installed.
    I do a lot of vinyl transfer work. I decided to upgrade sound cards to the E-MU 1212M to improve the overall quality of my audio.
    The card install went well. I'm receiving an audio signal just fine. I can playback at 44.1, 48, 96, 192.
    The only hitch is when I try to record. I can record at 44.1 & 48 but if I try to record at 96 or 192 I get the following message:"Another application has caused a sample rate change that makes the current session invalid. You may accept this change by loading session at the new sample rate indicated below...." The sample rate that it shows me is always 44.1 So when receiving this message it always wants to push me back to 44.1 This message pops up immediately after I hit the red "record" button.
    There are no other audio applications similar to Audition installed on my set up. I have turned off the system sounds, the 1212M card is recognized as the default audio device. The on-board sound card has been disabled along with its drivers. Gone also are the drivers from my previous sound card.
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    Ive also gone through what items are being loaded during start-up of my computer. Ive tried disconnecting from the internet and shutting down all programs that might be active, ensuring that no other programs are running other than Au 3 and the 1212M PatchMix panel.
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    -DLedin

    "Before you did that, did you start a new session at 96k, and it still won't record into it? "
    If you're asking if I first start a new session in PatchMix, yes, that's correct. I call up the 96k session in PatchMix first. Then I go and open up Audition to try to record.
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    If that is a what you mean by "session" then I would need to do some experimenting by setting PatchMix to 44.1, then open up Audition, hit "record", specify 96/24 in the window that pops up and see what happens. I would hope that I'd get a message from PatchMix that I need to go to set PatchMix for a 96k "session". That's actually a possibility I hadn't previously considered trying. I'm at work now but will look into this possibility after I'm back home & report back.
    -DLedin
    EDIT: Since my last reading of your message you added a question:"Also, when you set the session up, did you take the check out of the 'use default' box?"
    Please specify which application you're referring to when you say "session". PatchMix uses the term quite a lot. If there's a "use default" box somewhere that I need to take a look at then I need to know if it's in Audition 3 or PatchMix and just where to find it. Also, should that "use default" box be checked or unchecked? I appreciate your patience.

  • Z835 Sound card? Performanc​e?

    Sound is terrible on my Z835-ST8305.  Tried installing audio enhancement software from SRS.  Got message that there is no compatible sound card installed.  DFX installed but doesn't help much.  What IS my sound card?  Do I need new driver?  Any recommendations for audio enhancement software?
    Solved!
    Go to Solution.

    Are you talking about the sound on the built-in speakers, or on some plug-in external speakers, or on a set of headphones, or on all three? If it's only on the built-in speakers, you need to test the other two, so we can narrow down where the problem actually is. Because you're complaining about the volume, I'm thinking you're only talking about the built-in speakers. But, I've been wrong before, once or twice... <smile>
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  • Skype can't detect sound card, issue with playback...

    Hello
    I have just received my Razer kraken 7.1 headset and have installed razer synapse 2.0. I have set up my mic and speakers of the headset through the control panel, my playback is set as the razer kraken 7.1 earphone and the mic is set as the headset microphone through the recording tab. When I launch skype there is no sound produce through the application, however with previous headsets there has been, I have tried going on youtube to see if sound comes through the headset and it does. Previously I have been using a turtle beach z11 which used 3.5mm jacks in to the computer however this new headset is usb operated. When skype opens and I try to call someone it says problem with playback device meaning I can't call anyone at all. However when checking call quality it says everything is working fine even though it isn't.
    So far I have tried re-installing skype and restarting my pc none of this has worked so far, I know this isn't likely a fault with the headset as it works with all other programmes I have tried.
    I am using a custom built pc with
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    -Thanks
    George Snell

    @gsnell, Please setup your audio device directly in Skype and not only in the windows playback/record interface.
    Also, please try to quit synapse 2.0 since it may interfere with Skype.
    Also please try with the latest 6.13 version 
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    Thank you

  • I'm having a hard time with my sound blaster z sound card to sync with cs6. Can anyone share with me

    I'm having a hard time with Adobe CS6 synching up with my soundcard. Latency settings and other applications are challenging at this time. I had 3.0, but my Windows 7 computer is now "deceased". Please help.

    I endorse everything that Steve just said but to try and get you going while you're shopping for an audio interface more suitable for sound work than use as a doorstop, try the following:
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  • 5.1 or 7.1 external sound card

    I have a new iMac but it only has a 2.1 optical audio output on the back. I'd like to connect a 5.1 DD external sound card USB 2.0 (maybe even up to 7.1 Dolby Digital) to it to connect to my DD receiver. However, I am also using Windows Vista Ultimate through Boot Camp separately and may add VMWare Fusion later on to run both OSX and Vista side by side so I'd want to have both of them compatible and even before that. I've been looking for cards but haven't found any that are supported by both Apple AND PC. Either that or they are all internal PC cards. Does anyone know of any external cards that are supported by both?

    Tiery wrote:
    My iMac has a combination 1/8" optical/analog output jack on the back of it that
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    Radio Shack carries adaptors/cables of both types (usually at outrageously high
    prices). For online vendors, try http://monoprice.com Great prices and a good
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    Here are just two examples; browse monoprice.com for many more:
    [optical mini-plug to toslink adaptor|http://www.monoprice.com/products/product.asp?cid=104&cp_id=10423&cs_id=1042301&pid=2671&seq=1&format=2]
    [optical mini-plug to toslink cable|http://www.monoprice.com/products/product.asp?cid=102&cp_id=10229&cs_id=1022902&pid=1557&seq=1&format=2]
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  • Acquiring 24-bit sound with a third-party sound card... truncation issues.

    I am having a very strange problem when using LabVIEW to acquire audio
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    When capturing sound in 16-bit mode (as set in the LabVIEW software),
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    I have compiled a number of screenshots which showcase the problem in more detail.
    Here is a background of the experiment:
    For these tests, both the analog and digital audio was generated by an
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    respective line-level or digital audio inputs.  Digital audio was
    tested using both coax and optical cable.  In the sound card
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    into the WAVE IN L/R (via the Windows API, I assume).  The
    sampling rate for the profile was 96kHz.
    The sampling rate in all LabVIEW functions was set to 96kHz.  The sample rate set in the AP Digital generator was 96kHz.
    ANALOG 20dBu 96kHz 16bit.jpg
    In this test, everything looks fine.  The audio input is at
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    performance (with dithering).
    ANALOG 20dBu 96kHz 24bit.jpg
    Now we instruct the driver to capture sound in 24 bits.  Notice
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    ANALOG -20dBu 96kHz 16bit.jpg
    Now we drop the input level to -20dBu.   The performance
    starts to look a little messy but is still acceptable.  Note,
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    ANALOG -20dBu 96kHz 24bit.jpg
    Now we try to capture at 24 bits.  The effects of truncation are extreme at this low signal level.
    ANALOG -60dBu 96kHz 16bit.jpg
    Now we are at extremely low signal levels.  Individual
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    present.  Performance is still good.
    ANALOG -60dBu 96kHz 24bit.jpg
    However, when increasing the resolution to 24 (which should increase
    the number of quantization levels), our signal is reduced to a square
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    DIGITAL 0dB 16bit 96kHz 16bit.jpg
    Now on to the digital tests.  We start with full-scale.  We
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    cable.  The soundcard is instructed to receive in 16 bit
    mode.  It looks good.
    DIGITAL 0dB 16bit 96kHz 24bit.jpg
    Using the same input, we change to 24 bit receive mode. 
    DIGITAL 0dB 24bit 96kHz 16bit.jpg
    Now we set up the AP to output a properly dithered 24-bit signal at
    full-scale.  The dips in the frequency domain show us that
    something is wrong.
    DIGITAL 0dB 24bit 96kHz 24bit.jpg
    Receiving in 24-bit mode.  Same story as before.
    DIGITAL -90dB 16bit 96kHz 16bit.jpg
    Now we decrease the amplitude to a low level.  Well-implemented dither is shown here clearly.
    DIGITAL -90dB 16bit 96kHz 24bit.jpg
    However, receiving in 24-bit mode reduces the signal to a dithered square wave.
    DIGITAL -90dB 24bit 96kHz 16bit.jpg
    Here is the low-level signal with the AP generating a 24-bit
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    It seems the dither level has been changed.  This is the cause of
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    DIGITAL -90dB 24bit 96kHz 24bit.jpg
    And finally, we transmit and receive in 24-bits.  Here are the results.
    We have achieved similar results using several of your breakout boxes and soundcards.
    Attached are all screenshots, as well as the main VI (AP Test.vi) and
    the dependent vi's.  There are a number of SVT vi's in the
    project, but they can be ignored since they are not related to the
    problem.
    Any help would be greatly appreciated.
    Best Regards,
    Brett Gildersleeve
    Attachments:
    AP Test.zip ‏2381 KB

    Hi Brett,
    I took a look at the code you attached, but it appears that you may have left out the subvi that actually acquires the sound. When you say Windows API, I assume that you are calling a DLL at some point. How are you configuring the inputs (or what function are you using)? It could be (and this is just a guess) that when you specify 24 bits, the DLL returns the data in a very specific format. If you don't interpret the bits that are returned correctly, LabVIEW may not know what to do with them, as LabVIEW does not have a native 24-bit datatype. If there is any documentation for these function calls, our answer might there...
    Just some thoughts -- thanks for posting your solution!
    Charlie S.
    Visit ni.com/gettingstarted for step-by-step help in setting up your system

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