Multiple audio drivers / interfaces

logic 7.2 update manual states that 'Logic Pro supports plug and play for audio interfaces, making it possible to connect and switch on a new/additional audio interface while Logic is running. ..'
When I hit 'use' on the alert it swaps the driver - I can't see an option for running more than 1.
I am using MOTU 896HD and Mackie Onyx 1640 mixer with firewire option.
Both work independently just fine. What am I missing?
Thanks
Mana
PB G4 Mac OS X (10.4.9)

Bugs aside, if it's VERY noisy, that might just be the out-of-sync problem (something I'm very familiar with now).
Search the forum for "aggregate sync" and you should get some clues, if this is indeed the problem.
Short of it is, you need to sync the two devices together. Easiest way seems to be to connect the two devices via S/PDIF and set Clock Source to S/PDIF for the "slave" device (set in Audio Midi).
You can also just set the Clock Source as the "master" device in the drop down, bypassing the need for the S/PDIF cable altogether, but I haven't tried this myself.
Macbook Pro CD / iMac C2D Mac OS X (10.4.8)

Similar Messages

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.

Wont install audio drivers?!? Please help

My motherboard is the k7n2 delta series. I have 5.1 surrond sound speakers and I had just bought the S-bracket for this motherboard. I CANT get INTO the "NVIDIA nFORCE(TM) Control Panel" to configurate the speakers. When I install the drivers a warning comes up durring installtion:
The software your installing for this hardware:
NVIDIA(R) nFORCE(TM) Audio Codec Interface
has not passed windows logo testing to verify its compatibility with windows XP
continuing your installation of this software may impair or destabilize the correct operation
of your system either immediately or in future. microsoft strongly recommends that you stop installation
now and contact hardware vendor for software that has passed windows logo testing.
I cant get the NVIDIA(R) nFORCE(TM) Audio Codec Interface installed son then I cant configure the speakers so they are not working. Please help me. thankyou for your assiantace.

Hi,
Raven13601:
Read Here:
Guide to Realtek sound on NF2 boards
and here:
Nobby's Moan Guide
Realtek Drivers here:
http://ftp://210.51.181.211/pc/ac97/alc650/wdm_a356.exe
Axel

Audio/midi interface compatibility with Logic Express Mac osx 10.6.5

hi all, I'm looking for a compatible audio/midi interface to use with the latest mac os. I recently purchased a Ediroll UA25 but was unable to get it working . I installed the latest drivers from the Roland site but when plugged in it just wasn't being recognised. I sent it back and got a full refund but now I am insure if the Ediroll was faulty or if it wasn't compatible with os 10.6.5. I am now in the process of looking to purchase another interface but am looking for help from you guys.
Thanks in advance,
kiev

Hi,
just out of curiousity: did you use the UA-25 or the UA-25EX? I know the 25EX has a driver switch on it's back where you can enable a standard driver which should be compatible to OS X without an additional driver.
However, i didn't find another interface including a compressor except for the Yamaha Audiogram 6, if you need this feature.
If you don't need it, buy a MOTU, preferably with firewire, or a Presonus. If you want quality, maybe an Apogee Duet. I myself - due to shortage of funds - still use either my Tascam Fireone or my Alesis I/O 2. The Alesis sounds nice for it's price, but i won't buy any of these lower end interfaces anymore. You get what you pay for. I also used the Edirol UA-25EX for a time, but this was before snow leopard, so i can't really say "Yes or No", but it was definitely a very good interface compared to it's price. I won't recommend M-Audio, however, but others like it.
Fox

Home Theater need. Multiple Audio Streams from 1 card or possible to use 2 car

I currently have 2 SB Li've's installed and a SB Audigy in the box.
I would like to be able to send different audio out to different devices. I need to send audio from a game interface called MAME 32 to a speaker system connected to the PC. I also use this PC as a home theater PC and want to send audio from my DVD playing software out to my TV(in analog stereo mode) and/or to my receiver(in Dolby Digital mode using the digital connector).
I would like to be able to run these seperatly without having to go into the Control Panel and switch the default sound card like I am doing now. I do not need to run them at the same time.

Bummer. Are you guys working on something that might have this capability? Seems to me that the closer we integrate PC's into our entertainment systems the more need we will have for the ability to do multiple audio streams, i.e. MP3's to the living room. DVD movie audio to the family room, XM radio classical station to the kitchen for mom etc..
Intel is offering this as an onboard solution that has the ability to do at least 2 different streams.
http://www.intel.com/design/chipsets/hdaudio.htmMessage Edited by relay on 0-26-2004 05:02 PM

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

Adding Multiple audio tracks

Hi people.
How can I add multiple Audio tracks on a video which then can be switched on a DVD remote control?
Please help.
thanks in advance.
J.

jagthandi,
I agree with you on the separate purchase of Encore. It's a great authoring program. Until CS3, it was sold as a stand-alone, though was also included in the CS2 Production Studio, which is how I came by it. Adobe began integrating it more and more closely to Premiere Pro, so that became more of a "module," than a stand-alone. As of CS3 it only came bundled with P-Pro. Good on one hand, but not so good, on the other. Having used it for many years now, I would be loathe to use any other authoring application.
It seems that Sony has done similar with their DVD Architect Pro, only bundling it with Vegas Pro.
What this means is that there are now few full-featured authoring applications in an affordable price range. DVD Studio Pro being one of the few, and its interface is not as clean as Encore's. It offers potentially more power, but you have to do some basic programming to use that power. At the top of the list is Sonic Scenarist, but at around US$50K, complete, it's not really an option. Also, its interface takes years to learn.
So very glad that I have Encore2. I will likely never part with that program, as it does 99% of what I would ever want from an authoring application. That last 1% can usually be done with some creative workarounds.
Hunt

Audio/MIDI Interface Compatibility with Logic Pro 7.2

Hello,
Is/are there a recommended audio/midi interface(s) that will work without audio/midi timing glitches with Logic Pro 7.2? Currently, I am using Pro Tools LE v7.1, along with the DIGI002 Rack. That combination works very well, with no glitches whatsover. The possibility of using Logic Pro 7.2 raises the question of what would be a compatible interface. I don't think the DIGI002 Rack has the necessary drivers, although I may be mistaken. However, there must be an interface that works well with Logic Pro 7.2.
Any help or advice would be much appreciated.
Thank you,
TonyVT
iMac Intel Dual Core Mac OS X (10.4.6)

Hello,
I went to the Metric Halo Web site and was not able to find the interface that you may have been referring to. It looks like the Mobile I/O may be the correct one, however, that interface is much more than I would want to spend (http://www.mhlabs.com/metric_halo/products/mio/)
The price seems to be @ $1200. Anyway, I am somewhat confused regarding the Metric Halo.
Any additional advice/help would be appreciated.
Thank you,
Tony Ventura
www.tonyventura.com
iMac Intel Dual Core Mac OS X (10.4.6)

Audio/midi interface, FW800 enclosure, nearfield monitors for MacBook Pro?

I'm running Logic 9 on my MacBook Pro, having switched over completely to a laptop based solution from my previous Mac Pro.
My previous audio/midi interface was a Digidesign FW400 MBox, but I want to switch to a USB audio/midi interface plus an external FW800 enclosure as my recording drive.
I am also looking to replace my aging Roland nearfield monitors with something smaller. I mostly run soft synths and plug-ins (I have an old UAD-2 PCI and am trying to crossgrade to the ExpressCard solution too).
And I'm running Snow Leopard.
S0 my questions are these:
1) What would be a good USB2 audio/midi interface. I only need 1 x 1 for the midi (but 2 x 2 would be nice), plus XLR and 1/4 audio ins and both analog and digital outs. Decent Mac drivers a must!
2) Any recommendations for very small but decent quality nearfield monitors? I like the Genelec's, but for whatever reason they only appear to have XLR ins. SPDIF or 1/4 analog ins would be cool -- anything compatible with 1) above.
3) What is a good recommendation for a FW800 enclosure? Obviously looking for something quiet. My only audio in would be vocals, and they'd be recorded in the same room as the drives. There are some fanless options out there, but upon enquiry I've been advised I'd have to shut them down every four hours or so. I do have an old FW800 OWC twin drive enclosure, but despite the 'whisper quiet' claims, that thing sounds like a taxiing Learjet.
Thanks in advance for any thoughts or recommendations!

'Perfect' rather depends on what you want to do with it - microphone input? Level meter? Just a simple A-D converter? digital inputs?
I've used an earlier version of this device for some years with no problems (no MIDI though):
http://www.rolandus.com/products/productdetails.php?ProductId=758&ParentId=104
Other interfaces by Roland: http://www.rolandus.com/products/productlist.php?ParentId=104 - Edirol (taken over by roland) used to make MIDI interfaces as well but Roland don't seem to.
This page lists interfaces by several manufacturers:
http://www.bulletsandbones.com/GB/FAQPages/Interfaces.html

K8N Neo Platinum with Nvidia Unified Audio Drivers

Hi, I have two hard drives in my PC. I'm testing out the feasibility of me running X64 at the moment. I have newly installed X64 Windows XP on one new 2nd drive and used 5-5_xp64_dd_cp.exe for my ATI 9800 pro card with no problems. Then I have installed the nForce_6.25_WinXP64_international.exe audio drivers for my on board Realtek AC97 because i have heard they are better than the Realtek drivers themselves and the Nvmixer utility looks good. Ok, no problems.
I was so impressed with the X64 Nvidia drivers that I have decided in the meantime to replace the Realtek drivers with Nvidia ones on my 1st drive that has already got Windows XP Pro installed on it. So I uninstalled the Realtek and replaced them with these nForce_5.10_WinXP2K_WHQL_english.exe unified audio drivers.
But, for some reason, they dont work. As I understand, the Nforce 5.10 are unified drivers and should replace the original Realtek ones i had installed on my 1st drive. I see no speaker icon on my taskbar and when i click to open nvmixer on my programs list, a message appears stating it cannot open as there is no nvidia compatible sound hardware present in my machine. I already know this is not the case as the X64 Nvidia drivers work for my X64 installation on the 2nd drive.
Any ideas what i am doing wrong?

The sound quality on those compared to the 4.57 standalone nvidia (that has its own version of the realtek AC97 codec and updated interface drivers) is not as good. Theres a long thread on here if you search for it for peoples opinions.
Quote from: charliemor45 on 29-May-05, 20:32:56
Shovon, the latest AC97 drivers from www.msicomputer.com will work perfectly for the K8N Neo2 Platinum
I currently have thwm running and have no problems
Go to that website, put the cursor over the 'Support' header and a drop down menu appears.
Click on Downloads.
Scroll down till you see the mother board K8N Neo2 Platinum(under the main heading Slot 939/K8)
Click on the link, and you'll come to the drivers page.
The Listed AC97 drivers that are the latest are there.

Edirol Audio/Midi Interface and Yamaha Keyboard

Hi everybody,
I just got an Edirol Audio/Midi interface UA-20, installed correctly with latest drivers, garage band recognizes it. Now I want to connect my old Yamaha PSR SQ16 keyboard through it, but somehow it doesn't work. Can anybody please give me a guidance on how to connect it properly, maybe i'm too dumb for that...
Thanks for any help!

I'm shure you must have already figured this out, becouse it's very simple thing to do, but I'll post my answer anyway, perhaps someone else has the same problem.
The Yamaha PSR SQ16 has a "MIDI OUT" plug, in the back of the sinthetizer. To connect with de UA20 Edirol, all you need to do is to get a midi cable (you can find it in every music store) and plug the Yamaha PSR SQ16 MIDI OUT to the Edirol UA20 MIDI IN.
And boy, that's it.
All the fantastic sounds and samples that cames with Garageband, are now triggerd through your Yamaha PSR SQ16.
Things to check: in the AUDIO MIDI UTILITY, be shure that the UA20 is set as de midi in out device.
In Preferences of Garageband, be shure to check that the UA20 is set as audio and midi interface.
Hope this be helpfull.
I apologise for my english, it isn't my first language.
Saludos

Can anyone recommend audio/midi interface LE7 + Mac mini

Hi all!
I've been working with LE 7 for some weeks now. My setup is really simple: a Mac mini, LE7, some free plugs, that's it. Now, I've come to the stage when I need to record guitar, bass, keyboard and add some vocals. Could someone please recommend a decent interface that is compatible with my setup?! I don't have very hi-tech requirements, I just want something that works. No need for lots of inputs, and I've only got one keyboard. FireWire or USB should not matter, unless you know what to prefer. Low budget....
Many thanks in advance!

Firewire Interfaces are always better than USB ones for audio. I really love the Focusrite Saffire but maybe it's more than what you're willing to pay. You could also check out Presonus Firebox or M-Audio FW Solo. However, M-Audio drivers are sometimes a little on the buggy side. If you want to go the USB way, I'd check Tascam's US-122.

MSI 785GM-E51 can't install audio drivers

Hey, I would be grateful if you could help me with this problem.
When I try to install the audio drivers of this mobo(785GM-E51), Windows 7 can't find them. It offers me another set of drivers - that of VIA's, but it should be Realtek's. Normally it wouldn't be a problem, but I have a an issue with front panel microphone jack(it's not working), so I want to install the proper drivers. I tried disabling auto-installing option of Windows 7, and then install the drivers that MSI offers officially , but still the computer acts as if I didn't install anything.
EDIT: It seems that I have a different motherboard - in the document I have(about the parts of PC), it says that the computer has MSI 785GM-E51, but now that I checked the informationa about motherboard in the Windows itself, it appaars that ASrock M3A770DE is in the case. So I guess the fact that I can't install the MSI's drivers is because of different mobo. I have no idea how this could've happened.

>>Posting Guide<<
Why don't you open up a case and read the model? Is the computer secured with some sort of sticker that voids warranty if opened?

How to edit a multiple audio video clip on Premiere

Hello everybody
Maybe this is a dumb question, I'm sorry if I'm asking what is obvious for you, but I swear I tried to find on the web and on this forum, but I couldn't find anything.
Well, I made a video using a screen capture software called Dxtory, that provides me the ability to record my PC screen, with multiple audio tracks.
Ok, now I have a 15gb avi file. When I open this file in the Media Player Classic, I can play one or both channels. The first channel is the original audio captured from my PC. Every audio generated by my PC is in this track. The second track is my voice, with my commentaries. I have all of this in one single file. They are not splited. It's just one file.
Then when I import into Premiere, the software just recognize the video track and the first audio track. I can't find the second, with my voice.
I know I can split the clip into 3 files (1 video and 2 audios) but it would take longer. I just want to drop the file on Premiere and edit the audio volumes quickly, because when I talk I would like to low the volume of the PC audio.
Now, the question: Is there a way to Premiere recognize both tracks in this single file? How?
Thanks in advance for any help.

I'm actually surprised that Pr can import this at all--the video is Xvid. Maybe it's some variant that Pr's importers can handle, as Jeff suggested. I dunno--it doesn't play well for me, regardless, and I'm not going to install the Dxtory codec to find out.
Anyway, to the matter at hand: while AVIs can, apparently, contain multiple audio tracks, Pr's importers are limited to a single track. However, Pr has other importers than can handle multiple audio tracks. QuickTime--which is a wholly different process--supports multiple audio tracks, as does MXF (some flavors). However, Xvid in QuickTime (which is feasible) won't import in Pr (at least on a PC), and Xvid won't go into an MXF file at all; that means you'd have to transcode. Personally, this would be my choice--but I found that the original clip played back pretty terribly, so that would be why I'd go that route.
Additionally, you could extract the second audio track to a separate WAV file, import both the AVI and WAV, and then use the Merge Clips feature to marry them together as one pseudo-clip. Not perfect, but it would work. The benefit is that you don't re-encode anything.
So, I've got solutions for both the re-encode/MXF option (my preference) and the AVI/WAV option. Here's proof of the MXF (transcoded video to XDCAMHD422 50Mbps) with four audio channels (stereo must be split to dual mono):
At the end of the day, these (or a variation of them) are your only options. Pr simply won't import multiple audio tracks (even dual mono) in an AVI container. Let me know if you're interested in either of the solutions.

How do I import multiple audio tracks?

Hi there...
I have a number of video files that have multiple audio tracks in them. The problem is I can not work out how to get prem to see multiple audio tracks. The files import fine and I have also tried various other formates with multi-track audio.. but only audio "stream 1" is visible in prem.
What I was hoping is that the video would come in with a 2 audio tracks instead of 1 each in its own "lane".. so I can then use the mixer and the audio files are linked to the video file and in perfect sync.. (just like when you import with 1 track)
NOTE: I am not talking about channels.. but complete 2 completely different audio tracks.. and I know that I could extract these audio streams into wavs and then re-sync them but this seams overkill and a duplication of effort... surly there is a way to have prem support more than 1 audio track on a video file?
--Me

Well, if PP doesn't bring them in, then best guess is they just don't exist.
There's really no trick or user adjustment for this that I'm aware of. You import the file, PP sees what's in it. I've never observed or read about any other behavior than that.

Multiple audio drivers / interfaces

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