New ALC650 audio drivers

They fixed dodgy AC3 sounds in windows for me and the rear speakers now actually work in the speaker test  :D
http://www.realtek.com.tw/downloads/downloads1-3.aspx?lineid=5&famid=11&series=1&software=True

Quote
Originally posted by Rafterman
I do not use the onboard sound but AFAIK  imapi is a part of the Windows O/S, and has got to do with image mastering that applies to CD recording.  If you are using a burner then you probably do need it.
No, this is a new service.  The service that you are referring too is called "IMAPI CD-Burning Com Service".  That service is required (though it can stay in manual) and is part of XP and came from Microsoft.
However, after installing the new Nvidia audio driver, a new service was added, called "Imapi Helper".  This service is not part of XP, and did not exist on my system prior to installing this nvidia audio driver (I keep very close tabs on services and the like that run on my system, and this one was not there yesterday).
Furthermore, if you go into MSconfig Services tab, and choose to hide all microsoft services, this service does not get hidden.  It's not a Microsoft service, it was added by the Nvidia driver.

Similar Messages

  • New VIA Audio Drivers

    VIA has released a combo audio drivers to support all these southbridges :
    VT8231
    VT82C686A
    VT82C686B
    VT8233
    VT8233A
    VT8233C
    VT8235
    Here is the link : http://www.viaarena.com/?PageID=69
    And here is the direct download : (Windows) http://downloads.viaarena.com/drivers/audio/ComboAudio_a1u311b.zip

    I installed the MSI Drivers from the MSI cd. All ok but the clock and info tabs were missing. I upgraded the driver from msi site to 3000 and no tabs.
     X(  
    Before the GF4 i had a gf2 and i realized i forgot to uninstall the nvidia drivers before installing the new card's drivers. I uninstalled them and reinstalled the msi driver 3000. But no clock nor info tabs.
    Can someone help me? Are those tabs needed?
    I'm using xp pro, intel p4 1.8 on motherboard intel d850md, 1gb ram.
    Also the 3d turbo experience gives me msivga.ocx initialization error.
    Thanxs.
    Quote
    Originally posted by Assaf
    The MSI tabs only show if you use an MSI driver from the MSI website.
    They are nVIDIA drivers with the added MSI tabs and logos...
    But there is no real reson to do it.

  • New Nforce audio drivers?

    I noticed that nvidia have done a special release of just their audio drivers, has anyone tried these on the neo2, does it solve problems as they say it does?
    http://www.nvidia.com/object/nforce_audio_4.57_win2kxp
    there are no READMEs or instructions telling you how to install it.
    very strange.. probably a reaction to pressure from the half life 2 people..
    you can tell it has been done in a hurry because the "installation hints" tell you to uninstall your DISPLAY driver first  hehe.
    I guess it will give a similar benefits to that Bas described earlier when he did the same thing manually.

    Quote
    Originally posted by Rafterman
    I do not use the onboard sound but AFAIK  imapi is a part of the Windows O/S, and has got to do with image mastering that applies to CD recording.  If you are using a burner then you probably do need it.
    No, this is a new service.  The service that you are referring too is called "IMAPI CD-Burning Com Service".  That service is required (though it can stay in manual) and is part of XP and came from Microsoft.
    However, after installing the new Nvidia audio driver, a new service was added, called "Imapi Helper".  This service is not part of XP, and did not exist on my system prior to installing this nvidia audio driver (I keep very close tabs on services and the like that run on my system, and this one was not there yesterday).
    Furthermore, if you go into MSconfig Services tab, and choose to hide all microsoft services, this service does not get hidden.  It's not a Microsoft service, it was added by the Nvidia driver.

  • New M-Audio drivers

    Just got my M-Pulse email from M-Audio and they have some new drivers for some of their Firewire products to work with 10.5.2 http://www.m-audio.com/news/en_us-1267.html
    Of course let's how they work with 10.5.3.

    With important external hardware you must look before you leap with new computers or OS upgrades. This is the first time I've had an issue going to either with my M-Audio, so I don't blame them for taking a month + 1 week to release an update for a new computer.
    For example: When I have a chance I'll update the driver and test my M-Audio devices. If working, then I'll clone over my install to a firewire HD then install 10.4.8. Test the M-Audio devices again, then if it fails to work I'll go back to 10.4.7 via my clone and wait for a 10.4.8 driver.

  • A quick primer on audio drivers, devices, and latency

    This information has come from Durin, Adobe staffer:
    Hi everyone,
    A  common question that comes up in these forums over and over has to do  with recording latency, audio drivers, and device formats.  I'm going to  provide a brief overview of the different types of devices, how they  interface with the computer and Audition, and steps to maximize  performance and minimize the latency inherent in computer audio.
    First, a few definitions:
    Monitoring: listening to existing audio while simultaneously recording new audio.
    Sample: The value of each individual bit of audio digitized by the audio  device.  Typically, the audio device measures the incoming signal 44,100  or 48,000 times every second.
    Buffer Size: The  "bucket" where samples are placed before being passed to the  destination.  An audio application will collect a buffers-worth of  samples before feeding it to the audio device for playback.  An audio  device will collect a buffers-worth of samples before feeding it to the  audio device when recording.  Buffers are typically measured in Samples  (command values being 64, 128, 512, 1024, 2048...) or milliseconds which  is simply a calculation based on the device sample rate and buffer  size.
    Latency: The time span that occurs between  providing an input signal into an audio device (through a microphone,  keyboard, guitar input, etc) and when each buffers-worth of that signal  is provided to the audio application.  It also refers to the other  direction, where the output audio signal is sent from the audio  application to the audio device for playback.  When recording while  monitoring, the overall perceived latency can often be double the device  buffer size.
    ASIO, MME, CoreAudio: These are audio driver models, which simply specify the manner in which an audio application and audio device communicate.  Apple Mac systems use CoreAudio almost exclusively which provides for low buffer sizes and the ability  to mix and match different devices (called an Aggregate Device.)  MME  and ASIO are mostly Windows-exclusive driver models, and provide  different methods of communicating between application and device.  MME drivers allow the operating system itself to act as a go-between and  are generally slower as they rely upon higher buffer sizes and have to  pass through multiple processes on the computer before being sent to the  audio device.  ASIO drivers provide an audio  application direct communication with the hardware, bypassing the  operating system.  This allows for much lower latency while being  limited in an applications ability to access multiple devices  simultaneously, or share a device channel with another application.
    Dropouts: Missing  audio data as a result of being unable to process an audio stream fast  enough to keep up with the buffer size.  Generally, dropouts occur when  an audio application cannot process effects and mix tracks together  quickly enough to fill the device buffer, or when the audio device is  trying to send audio data to the application more quickly than it can  handle it.  (Remember when Lucy and Ethel were working at the chocolate  factory and the machine sped up to the point where they were dropping  chocolates all over the place?  Pretend the chocolates were samples,  Lucy and Ethel were the audio application, and the chocolate machine is  the audio device/driver, and you'll have a pretty good visualization of  how this works.)
    Typically, latency is not a problem if  you're simply playing back existing audio (you might experience a very  slight delay between pressing PLAY and when audio is heard through your  speakers) or recording to disk without monitoring existing audio tracks  since precise timing is not crucial in these conditions.  However, when  trying to play along with a drum track, or sing a harmony to an existing  track, or overdub narration to a video, latency becomes a factor since  our ears are far more sensitive to timing issues than our other senses.   If a bass guitar track is not precisely aligned with the drums, it  quickly sounds sloppy.  Therefore, we need to attempt to reduce latency  as much as possible for these situations.  If we simply set our Buffer  Size parameter as low as it will go, we're likely to experience dropouts  - especially if we have some tracks configured with audio effects which  require additional processing and contribute their own latency to the  chain.  Dropouts are annoying but not destructive during playback, but  if dropouts occur on the recording stream, it means you're losing data  and your recording will never sound right - the data is simply lost.   Obviously, this is not good.
    Latency under 40ms is  generally considered within the range of reasonable for recording.  Some  folks can hear even this and it affects their ability to play, but most  people find this unnoticeable or tolerable.  We can calculate our  approximate desired buffer size with this formula:
    (Sample per second / 1000) * Desired Latency
    So,  if we are recording at 44,100 Hz and we are aiming for 20ms latency:   44100 / 1000 * 20 = 882 samples.  Most audio devices do not allow  arbitrary buffer sizes but offer an array of choices, so we would select  the closest option.  The device I'm using right now offers 512 and 1024  samples as the closest available buffer sizes, so I would select 512  first and see how this performs.  If my session has a lot of tracks  and/or several effects, I might need to bump this up to 1024 if I  experience dropouts.
    Now that we hopefully have a pretty  firm understanding of what constitutes latency and under what  circumstances it is undesirable, let's take a look at how we can reduce  it for our needs.  You may find that you continue to experience dropouts  at a buffer size of 1024 but that raising it to larger options  introduces too much latency for your needs.  So we need to determine  what we can do to reduce our overhead in order to have quality playback  and recording at this buffer size.
    Effects: A  common cause of playback latency is the use of effects.  As your audio  stream passes through an effect, it takes time for the computer to  perform the calculations to modify that signal.  Each effect in a chain  introduces its own amount of latency before the chunk of audio even  reaches the point where the audio application passes it to the audio  device and starts to fill up the buffer.  Audition and other DAWs  attempt to address this through "latency compensation" routines which  introduce a bit more latency when you first press play as they process  several seconds of audio ahead of time before beginning to stream those  chunks to the audio driver.  In some cases, however, the effects may be  so intensive that the CPU simply isn't processing the math fast enough.   With Audition, you can "freeze" or pre-render these tracks by clicking  the small lightning bolt button visible in the Effects Rack with that  track selected.  This performs a background render of that track, which  automatically updates if you make any changes to the track or effect  parameters, so that instead of calculating all those changes on-the-fly,  it simply needs to stream back a plain old audio file which requires  much fewer system resources.  You may also choose to disable certain  effects, or temporarily replace them with alternatives which may not  sound exactly like what you want for your final mix, but which  adequately simulate the desired effect for the purpose of recording.   (You might replace the CPU-intensive Full Reverb effect with the  lightweight Studio Reverb effect, for example.  Full Reverb effect is  mathematically far more accurate and realistic, but Studio Reverb can  provide that quick "body" you might want when monitoring vocals, for  example.)  You can also just disable the effects for a track or clip  while recording, and turn them on later.
    Device and Driver Options: Different  devices may have wildly different performance at the same buffer size  and with the same session.  Audio devices designed primarily for gaming  are less likely to perform well at low buffer sizes as those designed  for music production, for example.  Even if the hardware performs the  same, the driver mode may be a source of latency.  ASIO is almost always  faster than MME, though many device manufacturers do not supply an ASIO  driver.  The use of third-party, device-agnostic drivers, such as  ASIO4ALL (www.asio4all.com) allow you to wrap an MME-only device inside a  faux-ASIO shell.  The audio application believes it's speaking to an  ASIO driver, and ASIO4ALL has been streamlined to work more quickly with  the MME device, or even to allow you to use different inputs and  outputs on separate devices which ASIO would otherwise prevent.
    We  also now see more USB microphone devices which are input-only audio  devices that generally use a generic Windows driver and, with a few  exceptions, rarely offer native ASIO support.  USB microphones generally  require a higher buffer size as they are primarily designed for  recording in cases where monitoring is unimportant.  When attempting to  record via a USB microphone and monitor via a separate audio device,  you're more likely to run into issues where the two devices are not  synchronized or drift apart after some time.  (The ugly secret of many  device manufacturers is that they rarely operate at EXACTLY the sample  rate specified.  The difference between 44,100 and 44,118 Hz is  negligible when listening to audio, but when trying to precisely  synchronize to a track recorded AT 44,100, the difference adds up over  time and what sounded in sync for the first minute will be wildly  off-beat several minutes later.)  You are almost always going to have  better sync and performance with a standard microphone connected to the  same device you're using for playback, and for serious recording, this  is the best practice.  If USB microphones are your only option, then I  would recommend making certain you purchase a high-quality one and have  an equally high-quality playback device.  Attempt to match the buffer  sizes and sample rates as closely as possible, and consider using a  higher buffer size and correcting the latency post-recording.  (One  method of doing this is to have a click or clap at the beginning of your  session and make sure this is recorded by your USB microphone.  After  you finish your recording, you can visually line up the click in the  recorded track with the click in the original track by moving your clip  backwards in the timeline.  This is not the most efficient method, but  this alignment is the reason you see the clapboards in behind-the-scenes  filmmaking footage.)
    Other Hardware: Other  hardware in your computer plays a role in the ability to feed or store  audio data quickly.  CPUs are so fast, and with multiple cores, capable  of spreading the load so often the bottleneck for good performance -  especially at high sample rates - tends to be your hard drive or storage  media.  It is highly recommended that you configure your temporary  files location, and session/recording location, to a physical drive that  is NOT the same as you have your operating system installed.  Audition  and other DAWs have absolutely no control over what Windows or OS X may  decide to do at any given time and if your antivirus software or system  file indexer decides it's time to start churning away at your hard drive  at the same time that you're recording your magnum opus, you raise the  likelihood of losing some of that performance.  (In fact, it's a good  idea to disable all non-essential applications and internet connections  while recording to reduce the likelihood of external interference.)  If  you're going to be recording multiple tracks at once, it's a good idea  to purchase the fastest hard drive your budget allows.  Most cheap  drives spin around 5400 rpm, which is fine for general use cases but  does not allow for the fast read, write, and seek operations the drive  needs to do when recording and playing back from multiple files  simultaneously.  7200 RPM drives perform much better, and even faster  options are available.  While fragmentation is less of a problem on OS X  systems, you'll want to frequently defragment your drive on Windows  frequently - this process realigns all the blocks of your files so  they're grouped together.  As you write and delete files, pieces of each  tend to get placed in the first location that has room.  This ends up  creating lots of gaps or splitting files up all over the disk.  The act  of reading or writing to these spread out areas cause the operation to  take significantly longer than it needs to and can contribute to  glitches in playback or loss of data when recording.

    There is one point in the above that needed a little clarification, relating to USB mics:
    _durin_ wrote:
     If  USB microphones are your only option, then I would recommend making  certain you purchase a high-quality one and have an equally high-quality  playback device.
    If you are going to spend that much, then you'd be better off putting a little more money into an  external device with a proper mic pre, and a little less money by not  bothering with a USB mic at all, and just getting a 'normal' condensor  mic. It's true to say that over the years, the USB mic class of  recording device has caused more trouble than any other, regardless.
    You  should also be aware that if you find a USB mic offering ASIO support,  then unless it's got a headphone socket on it as well then you aren't  going to be able to monitor what you record if you use it in its native  ASIO mode. This is because your computer can only cope with one ASIO device in the system - that's all the spec allows. What you can do with most ASIO hardware though is share multiple streams (if the  device has multiple inputs and outputs) between different software.
    Seriously, USB mics are more trouble than they're worth.

  • Windows 7 Wireless and Audio Drivers on 2010 MacBook Pro

    I have a new MacBook Pro i7 2.66 Mhz with Snow Leopard and dual boot to Windows 7 64 bit Ultimate. I can't find the correct drivers (Windows 7 side) to get the audio and the wireless networking operating. I ran the bootcamp setup from the Snow Leopard install disc on Windows 7. It gave an error message that this model is not supported. I then ran each software driver individually from the install disc and was able to get all to work except the audio and wireless. I then installed the Bootcamp 3.2 update from the Internet with no improvement.
    Does anybody know where the audio and wireless drivers can be located for Windows 7? (Apple Support has not been able to resolve the problem yet) Thank You!

    I'm having the same problem with Windows 7 64-bit Professional. I had to go through the same process to get most of the drivers installed. I've had the same issues with the wireless and audio drivers.
    To get the Apple drivers installed, I followed the instructions from this blog post.
    http://michael.anastasiou.me/index.php?option=com_content&view=article&id=22:qbo ot-camp-x64-is-unsupported-on-this-computer-modelq-solution&catid=7:how-to&lang= en
    Has anyone had any luck with the rest of the drivers?

  • Apple's Core Audio drivers have been corrupted?

    Since upgrading to Leopard, my Alesis USB Multimix8 mixer has been increasingly getting more and more static output and has become about useless. After trying everything I could think of, I wrote to Alesis and this is what their support said...
    "With Apple’s latest updated OS Leopard, it seems Apple’s Core Audio drivers have been corrupted and affects many class compliant devices such as the Multimix 8 USB which relies on these drivers. Until Apple addresses this with a fix, the best workaround is to back-down to 10.4.10, especially if this will be used in a professional studio."
    Does this sound correct to anyone? Is theAudio Core drivers that bad on Leopard and is this something that Apple even plans to address?
    Thanks!

    Hi, everyone.
    Alesis is VERY quick to point the finger at Apple for all their woes, but after my last experience, I'm thinking this is Alesis' problem.
    At one time I was running my Alesis MultiMix8 FireWire and getting constant kernel panics while working on audio and video deadlines. As this was the famed 8-core (rolling off the assembly lines in May and all that) I was banging my head against the wall. It finally came down to me bringing in all of my office into Apple's Genius Bar at Tysons Corner, VA and trying to duplicate the occurrence.
    The culprit was the Alesis driver.
    When I upgraded to Leopard and the new Alesis FireWire driver, I have encountered no KPs or odd audio issues. Confident, I decided to try out the Alesis iMultiMix8 USB for the Second Edition of Podcasting for Dummies. I got a high-pitched whine/static when I tried working with the audio mixing between iPod and mixer. I returned the mixer and exchanged it for another one. Same problem. I then went and tested the iMultiMix on my PowerBook G4 running 10.4.11. Problem still existed. I wrote to Alesis and told them about the iMultiMix, asking them if this was a common issue.
    No response.
    While Leopard has solved the issues of the KP and continues to perform like a champ, I have been talking to other friends working on both 10.5 and 10.4 having issues with Alesis mixers and USB-powered pre-amps. It sounds less of an Apple issue and more of an Alesis issue.
    I think all we can do is wait for Alesis to resolve the problem they are failing to acknowledge is theirs. Not much of an answer, but apparently this the way things appear to be.

  • Switching Core Audio Drivers - No Audio Out

    I have a project in Logic Pro 8. It was recorded with Presonus Core Audio drivers on a Powerbook. The Project was then handed off to me to complete here in the studio.
    I re-opened the Logic Express Project in Logic Pro 8 using Digidesign HD hardware and Digidesign's Digidesign HW (HD) Core audio driver. All was well.
    We decided to try some changes in routing, using some new equipment M-Audio (ProFire Lightbridge) to take advantage of some Apogee I/O units in the studio. Loaded up the M-Audio Core Audio Drivers (ProFire Lightbridge Multichannel), made sure everything appeared properly in the Apple AMS (Audio Midi Setup) panel and did test tracks in Logic. All was well still
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    I rechecked the test project with tones. I/O activity normal. Audio out. I opened another of the series of sessions from Logic Express to Logic using DAE now using the M-Audio interface. Nada. No firewire/ADAT activity?
    I have reverted without problem back to using the DAE/Digi Core Audio model with no problems, but I would like to get the ADAT/Core Audio model running on these same projects as a proof-of-concept, and see no reason why it shouldn't.
    Opinions, suggestions, ideas, questions?
    ...JB
    Message was edited by: J A Bohrer

    When I was on the phone with M-Audio Tech Support (they're quite helpful) we noted the following:
    1. If you switch over to Core Audio using the Mac's Internal speaker in one of these projects (switch to Apple's internal Core Audio Driver). All Channels are immediately updated to show only Output 1-2 available. and the project plays through channels 1-2 and the MacPro Speaker.
    2. Switching back to DAE Core Audio, all available channels are updated and all plays/records normally using the Digidesign HW (HD) Core Audio Driver.
    3. When the ProFire Lightbridge Core Audio Driver is selected, all channels are updated (now 1-18) and available, but if you select a channel or channel pair on one of the Channel Strips that was not in use previously in this session, an "out" bus for that channel is not created in Logic (which it normally does, assuming you choose a new channel pair for output from a channel strip...)
    When a 'new' project is created with the Lightbridge setup selected as the Core Audio output, all channels will auto create an appropriate channel output strip.
    My problem only exists when opening a previously created project. (In this case using DAE Core Audio...and I'm not implicating that as the cause, I don't think it is.)
    It's like having previous sessions/projects, buying new hardware to improve your ability to work with them only to find out you can't use your new stuff with the old sessions. (Not like that hasn't happened before.)
    Either Logic is at fault, not being able to accomodate the driver change from DAE to M-Audio, or the M-Audio driver is, being unable to get Logic to update its Environment to accomodate the new channel availability.
    ...J
    Message was edited by: J A Bohrer
    Message was edited by: J A Bohrer

  • Garage band wont work, need to install midi or audio drivers?

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    Im having the same problem, I upgraded my IMAC and was using tiger and I upgraded it to MAC 10.5.7, now GarageBand wont work saying my midi and audio drivers arent correct. My Reason 4.0 wont start either saying the same thing failed to open the midi library. My Protools works fine but I dont know how to update the audio and midi drivers that it matches the software I uploaded. I also went to the audio midi set up and it read, Audio MIDI Setup was unable to launch the MIDI Server. This means that you will be unable to view or edit your MIDI configurations.
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  • Garage Band Audio Drivers

    After Upgrading my OS to 10.5.8 from 10.4.1 I am unable to run Garageband '08. When I try to play one of my old projects the following message appears:
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    I did download and installed PACE drivers, which I believe allow you to open protected files.
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    RL

    Thanks for the prompt reply  Cristoph. That is exactly what I did. It turns out there were a few prior discussions
    regarding this issue, all advocating for the same method:  I went to the Library folder,  took out both Digidesign drivers from the Audio Midi Folder created a new folder within the Library and dumped them into it.
    Voilå!! Garage Band works again! Thank you.

  • Possible conflict between one or more third party midi or audio drivers

    just wondering if anyone has any new info about this error yet. i can't resolve it and still can't open many of my logic 7 files.
    thanks

    koltonmusic wrote:
    thanks for responding.
    i have many times removed and or re-installed all of the midi drivers.
    this problem occurs even without any midi or audio drivers in the system. this has been a problem since logic 8 came out for me and others. one user sent some L7 file sto apple and he said that evn apple support has reproduced the problem. that was last year.
    i was just hoping that there might be some new info out there.
    thanks
    Hi,
    In this case, I recommend you do a clean install of your operating system. Make sure you back up your important data beforehand.
    Afterwards, install Logic 8, and test it with the internal audio only. In other words do NOT install any third party drivers of any kind.
    This should work very well. If it does not, then I would suspect your motherboard has issues, and should be replaced by Apple.
    Cheers

  • New Audigy Beta Drivers did they add EAX 4.0 to an Audigy 1 Platinum

    Hello,
    does the new Audigy Beta Drivers add EAX 4.0 Support to an Audigy Platinum ?
    Greetings Picard007

    Audigy (hardware base one) supports EAX3 (EAX Advanced HD) hardware. Audigy2 suports EAX4 hardware. EAX4 is supported with latest OpenAL drivers on compatible hardware.
    More: http://en.wikipedia.org/wiki/Environ...dio_extensions
    Quote:
    EAX has nothing to do with actual 3D audio positioning. Positioning is done by Microsoft's DirectSound3D API. An alternati've to DirectSound3D, called Open Audio Library (OpenAL), surfaced in 2003 in several titles. OpenAL allows direct hardware acceleration of audio, like DirectSound, including EAX. As of 2006, the API has been used in many popular titles including Doom 3 and Prey. These games support EAX 4.0 if audio hardware with an OpenAL-supporting driver is present.
    Most releases of EAX versions coincide with increases in the number of simultaneous voices processible in hardware by the audio processor: the original EAX .0 supports 8 voices, EAX 2.0 allows 32 (Li've!), EAX Advanced HD (EAX 3.0) supports 64 (Audigy), EAX 4.0 again supports 64 (Audigy 2), and EAX 5.0 allows 28 voices (and up to 4 effects applied to each) (X-Fi).
    Message Edited by SoNic2367 on 0-07-2006 02:0 PM

  • Realtek AC97 Audio Drivers Not Working!

    Hi.
    I recently had a HD crash and I was forced to format it. I installed everything just as the first time, and everything worked fine except the audio drivers. The problem is, it does actually install the drivers, but when I look at Sound Options or Device Manager there are no drivers for the soundcard. It says installation is complete and then I restart the computer, and when I try to update with MSI Live Monitor 3, it says I got the latest drivers installed. STILL doesnt work.
    Does anyone have a hint about the problem? Any virus? On another note Norton discovered viruses on my computer, even though I had installed the antivirus-software before connecting to internet\network. This is what it found:
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    Quote
    motherboard broken??
    Possibly. One way to try is see if it works fine with different os'es. If you can download knoppix's latest version (which I hope supports the onboard audio, like my new fedora linux does) and try it, then you can find out if the sound works or not.
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  • No Volume in system tray, master volume disabled, no audio drivers - Dv6500 model

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    I can't help, other than to say this happened to me today with the same hardware, only with GarageBand.
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  • [Vox] MSI VOX TV BOX USB 2.0 problem with audio drivers compatibility

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