Core Audio & Drivers

"Mac OS X includes USB and FireWire class drivers for connecting audio and MIDI devices to your Macintosh. Mac OS X will automatically recognize many devices — those that comply with the industry standard specification for MIDI and audio devices — when you plug them in. You don’t have to install extra software to use them."
This is taken fron Mac OS X / Core Audio web page.....
my question is....and the correct iSub (USB plugged) Drivers?where are they gone?
the iSub doesn't work properly since 10.4!!
what can we do?

a -39 is an End Of File error, which unfortunately does not really identify which file is corrupt in GB
the first thing i would try is to rescue the recordings (hoping that it's the GB file itself that is corrupt):
http://www.bulletsandbones.com/GB/Tutorials.html#allaboutrescuinggaragebandfiles

Similar Messages

  • 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
    Then I opened the Project, which had been working in Logic 8 and interfacing with the Digi HD hardware, now using the Lightbridge Multichannel driver to access the Apogee I/O, and...no audio.
    No indication of activity on the hardware interfaces. I checked Logic 8, onscreen everything appeared normal. Channel meter activity. Channel routing remained unchanged and should have appeared at the appropriate D/A convertor. Digital activity which would have normally been indicated at the ADAT to Firewire interface was non-existent. So, I would deduce that the computer was not directing the Logic channel activity through the new driver to the firewire interface from this project.
    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

  • Core Audio Drivers not Found Error (no Logic Pro Installed)

    I am receiving the following error in GarageBand 08 and was hoping someone might be able to assist.
    When recording with a USB mic in GarageBand 08 (MXL USB.006) I sometimes get the following error: (Core Audio Drivers not Found). The error occurs after I have completed a recording session and removed the USB mic. (The error occurs regardless of if I close GB before removing the mic; but only occasionally)
    In checking the GB preferences, the Built-In audio option has disappeared from the drop down menu entirely for both Audio Input and Audio Output (although it still can find the USB mic if I reattach). In the Utilities folder, audio preferences, the Built-In audio option is also missing from the drop downs. (Not grayed out -- missing).
    In researching the problem, it looks like a similar problem occurs with the Logic Pro suite but I don't have that installed on my Mac (and never have). In contacting MXL support, they noted that it was a problem with Leopard and they didn't have a fix.
    At this point, I'm looking for either (1) a fix I've missed in my research or (2) an easy way to reinstall/replace/find the drivers without having to restore my system from backup. I've received the error three times now and restoring from back up - which clears the error and restores the drop down - is getting tiresome.
    Thank you!

    Thank you for the suggestion! I believe that was the issue the first time I lost the drivers. Unfortunately, even resetting the inputs and closing GB before disconnecting doesn't guarantee my Built-In audio won't disappear. Thank you for trying though!

  • Core audio drivers returned an error code -39

    hi
    when i was trying to export a song garageband begins mixing down then returns "error code -39", anyone know whats going on?
    thanks

    a -39 is an End Of File error, which unfortunately does not really identify which file is corrupt in GB
    the first thing i would try is to rescue the recordings (hoping that it's the GB file itself that is corrupt):
    http://www.bulletsandbones.com/GB/Tutorials.html#allaboutrescuinggaragebandfiles

  • Can I use a Protools HD interface as core audio for Logic 9?

    I am moving to a studio that already has Protools HD running on Mountain Lion.
    I don't especially like PT, so my question is... can I just use the PT HD hardware already there and run Logic and my AU plugs without using TDM?
    And if I do, how would it compare to the RME Fireface 800 I am currently using?
    Or would I be better off taking the RME and switching between the units depending on who was driving in the studio?
    Thanks in anticipation

    Yes, higher latency...
    Avid/Digidesign never put a lot of effort into their core audio drivers because they would rather have you purchase their hardware which uses proprietary drivers.
    There is an I/O limit but from the looks of it you won't be coming close.
    Offhand I'd say you will be missing the RME hardware but it may work for you.
    Keep us posted as to how it's going.

  • Logic pro 9  quit unexpectedly while initializing Core audio

    i've installed my logic studio 9 and it does not work, it quits unexpectedly while initializing Core audio I am using mbox 2 mini but i don't like protools, i deleted it. I've installed the digidesign standalone core audio drivers 8.0.3 which is compatible with my OS, but logic pro doesnt launch, soundtrack pro works and waveburner does too, mainstage and logic pro do not. I've configurated system preferences/sounds and mbox2 mini appears right. I've configurated applications/utilities/audio midi setup and everything is correct , system preferences/ mbox2 says "connected to mbox 2 mini" so if everything is correct why it doesnt work in logic pro and with soundtrack works perfectly? OMG it drives me nuts already... i tried also run logic pro without audio interface, just with "built in" device. it didn't work too. i launched it pressing control(without launch core audio) in audio settings appears mbox2 mini but doesnt work. I did read in a forum "you should delete logic's preference file" .... didn't work too.... like i said can't deal anymore with this issue. please HELP!!

    i've installed my logic studio 9 and it does not work, it quits unexpectedly while initializing Core audio I am using mbox 2 mini but i don't like protools, i deleted it. I've installed the digidesign standalone core audio drivers 8.0.3 which is compatible with my OS, but logic pro doesnt launch, soundtrack pro works and waveburner does too, mainstage and logic pro do not. I've configurated system preferences/sounds and mbox2 mini appears right. I've configurated applications/utilities/audio midi setup and everything is correct , system preferences/ mbox2 says "connected to mbox 2 mini" so if everything is correct why it doesnt work in logic pro and with soundtrack works perfectly? OMG it drives me nuts already... i tried also run logic pro without audio interface, just with "built in" device. it didn't work too. i launched it pressing control(without launch core audio) in audio settings appears mbox2 mini but doesnt work. I did read in a forum "you should delete logic's preference file" .... didn't work too.... like i said can't deal anymore with this issue. please HELP!!

  • 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.

  • The core audio for Logic 8 has stopped working.

    I was successfully running Logic 8, Protools LE and Cubase on a MacPro with Leopard (10.5.5) and a MOTU traveler interface. I loaded on the Ignition Pack bundled with Protools LE (Mbox mini), including Melodyne/Reason/Live 6. This then stopped me opening any of my programs, Logic, Protools, Cubase and Garageband.
    I have surfed several forums and found that Melodyne has caused the same problem for others. Following advice given on these forums I have...
    - Deinstalled all new programmes, including all links, files, rewire stuff
    - I have trashed Logic preferences
    - I have completely reinstalled Logic
    - I have updated the latest MOTU drivers and Apple software updates
    I can open Logic using Ctrl and selecting 'no' to core audio, but when I change the core audio, either to MOTU or built-in, and enable Logic freezes and I have to force quit. I have read other threads with people having the same problem, but after following all the advice the problem is still there. Help!
    Just for info: I cannot not open Cubase, Protools or Garageband, but there is no problem with standalone Kontakt 2, Itunes, Soundtrack Pro, Waveburner or Mainstage, using the MOTU.
    Thanks in advance for any help and advice.
    Mac Pro, 2 x 3 Ghz Dual Core Intel Xeon, 5 GB RRAM, OS X (10.5.5), MOTU Traveler, Mbox mini

    jmallalieu wrote:
    I was successfully running Logic 8, Protools LE and Cubase on a MacPro with Leopard (10.5.5) and a MOTU traveler interface. I loaded on the Ignition Pack bundled with Protools LE (Mbox mini), including Melodyne/Reason/Live 6.
    Sounds like a problem with an incompatible version of either Melodyne or Rewire.
    Rewire is used to connect application. Ex. Reason can run rewired into Logic, the audio output of Reason will appear as a separate audio channel in Logic.
    Two choices... see if your version of Rewire can be upgraded to the latest version. (don't know if that's even compatible)
    Remove Rewire from your system, move the rewire plugin out of it's folder, to the desktop. (run a search)
    The other plugin that might be giving you problems is Melodyne, move the plugin to the desktop. Again, run a search to find this stuff.
    pancenter-

  • How do I use 2 Core Audio devices at the same time in Logic w/out switching

    Hello, I use the Aardvark Direct Mix usb3 as my audio interface for mic and guitar. I just upgraded to Logic Pro 7 today and was hoping someone could tell me how I could use the Built-in Audio Core Audio driver along with the Direct Mix USB Core Audio Driver at the same time without having to switch Core Audio Driver's and than rebooting Logic. It is very frustrating doing that as well as not being able to hear my Logic audio during playback/record mode when I am trying to record via the USB3. WHat can I do? Thanks! If ther are both core audio, why can I not just use both of them at the same time? Why must a switch back and forth? Thanks!
    [email protected]

    THanks everyone! awsome article! Thanks for posting it, nh!
    The weird thing is, once I go up to the Menu and select Open Aggregate Device Editor in the AUDIO/MIDI setup utility and try to add a new Aggregate Device setup (step # 6 from the link), the only audio device available is the "Built-in Audio". What's weird is when I have the main Audio MIDI Setup utility open, every single drop-down menu/tab has the "Direct Mix USB 3 1.0.9.1" device right beneath the "Built-in Audio" option. To me that says that the core audio device is indeed recognized by my computer (obviously since I can record with it using Logic). Can anyone help me with this one? Any suggestions?
    Not that it should matter but, the hardware audio interface that I am using is made by the no-longer-existant company once known as Aardvark. I actually had to purchase a third party Core Audio driver just to have it recognized by OS X.
    This is the website where I purchased my audio driver :
    http://www.usb-audio.com/
    A little more on how this device and it's core audio driver function within Logic . . . (Hopefully this might clarify something I may be screwing up) . . .
    I launch Logic with its default Built-in Audio which allows playback of audio from the app. I than go up to Audio Hardware and Drivers and than drop the flip down Driver menu down to the second option, "Direct Mix USB 3 1.0.9.1." Logic than prompts me that I should try to (re)launch Logic Pro to load the appropriate CoreAudio.. I do that. Now I am ready to record guitar through the Direct Mix USB3, I just can't hear anything (unless I use head phones as an output from the Direct Mix).
    Anyway, you all probably already knew what I meant so explaining that was pointless. But my question is, if the Driver is obviously recognized by my comp in both the Audio MIDI Setup and in Logic, than why won't the Direct Mix device show up as a possibility for setting up an aggregate device? Any suggestions? Thanks!
    Power Mac G5 Dual 2.7Ghz 4.5gigs of RAM   Mac OS X (10.4.7)  
    Power Mac G5 Dual 2.7Ghz 2.5gigs of RAM   Mac OS X (10.4)  

  • How do I find and turn Core Audio On.

    When I try to open Logic it stops at “Initializing Core Audio”.
    I have read the topics page and carried out all the suggestions including starting with the control key down to get into logic. When I do this logic opens and a box comes up with Launch Core Audio Driver, Yes or No. If yes, it freezes, if no I can load it up, but there is no core audio at all when I do.
    In the inspector, the CORE AUDIO is greyed out. I have gone to Preferences and ticked the Core Audio box and ticked “Apply” and then it freezes again. Logic cannot see or find the Logic Core Library.
    How do I turn Core Audio on? I have No Sound at all. There is no instrument list ( EXS 24, ES1, ES2, Sculpture, Ultrabeat etc) coming up in the I/O box, it only says No Plugin. When I load a previous song there is a line through the I/O instrument and no sound. In the arrange window the audio levels show that there is activity but there is no sound and the instruments are crossed out.
    It seems like there is a communication problem between the program and the Core Audio sound banks. Could this have something to do with the drivers? If so, Where can I get the drivers from and how do I put them in.
    Does anyone know how to get inside Logic maybe to a lower level so that I can tick some box to get the sounds back again.
    If I re-install the program will it fix the problem?
    Sorry to be so long-winded but I am in a mess with this, so any kind of advise would be very much appreciated.
    Many thanks.
    Roy.

    I haven't had this problem, but on other computers, I have seen some sound functions only work if you don't run certain other things on start-up. Try opening System Resources:Accounts:<your account>:Login Items, and clearing out the list.
    Hope you solve it!
    DS9

  • Help needed, can no longer insert Battery3 as a Multi, Core Audio Error.

    Hi all,
    Well I have been using a few different Motu interfaces as of late, the Ultralite, Traveler and now the 828 MK2. Each device has had no troubles with hBattery as a muti untill now the 828.
    I have loaded all sorts of other mutli devices but for some reason battery is a no go, I get a core audio error of -1 ?
    Now I have yet to install a different driver set since it was my understanding all Motu drivers are the same and the Audio interface driver page shows the 828 so I figured all was well.
    Any one have any ideas as to what I can do? This one has me stumped.
    Thanks in advance!

    I have the exact same issue, battery 3 won't load as a multi. it's authorized and works as a stereo audio unit, but when i try to load the multi, i also get error code -1. btw, this happened today - yesterday the multi worked. i already trashed the logic and battery prefs, no go.

  • Core Audio: Selected Driver not found. (-10202) error message and speakers don't work

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