Mixer help.

Hi everyone. I have a Korg D1600mkII mixer from a while back and noticed that I could plug it into my mac. I want to plug it in and use it my previous songs stored in the mixer so that I can individually edit each track. will garageband even read it? If so, can I individually edit each previously recorded track? thanks in advance.

You'll probably need to send the mixer audio output through a USB audio interface to your mac/BG. Then play 1 track at a time while recoding it to BG. Eventually you'll have each mixer audio track on it's own BG audio track and then you can work from GB.
I don't think there's a way to easily transfer audio data via USB from an external device directly into GB.

Similar Messages

Basic Mixing Help Please

Hi,
I can't even pretend Im all that new to Logic but I have some questions about Mixing particularly the channels and how they are set up.
Im working on a song at the moment. There are two things confusing me:
1: When I solo a track, an audio track, another track seems to plays through it. In this case its a percussion loop thats play over it.
Im not sure how I did this. I also note that Audio tracks are normally colored orange and instrument tracks blue. But the main vocal
track appears to be blue even though its an Audio track used to lay vocals down. Am I sending two signals through one track somehow? Im really bad at this sorry.
2: I cannot see a master channel, normally its right down the right hand end of the mixing table. Id use it to add some mastering effects to give the song some kick! But its not there. Ive tried singling out the master by hitting ALL, Single, Audio etc...But cannot see it.
Im sure two problems ahead are connected and down to my naivety with Logic.
Any easily digested help greatly appreciated!
Thanks

Yeas the Indian Park is the offending channel. It appears to play through other channels when you solo them, as does one of the AUDIO tracks, it plays on all other tracks when u try solo them. This Audio track is the one I spoke about earlier. I notice the Indian Park and Audio Channel appear to have the Solo button with a RED line through it?? Anyway Il send on some screen shots now, thanks I think the mix is Stereo.
Arrange Snap Shot
Offending Track selected with inspector window
You notice the Solo button with Red Line through it on Audio track 4 and indian Park??
And why is one of the Audio tracks blue?? Track Number 3??
And Finally, Arrange window only
Thanks for your patients on this, really appreciate your help!

Multiple input device recording/mixing help in Audition 5.5

I'm presently evaluating Audition 5.5 and haven't been able to find any information pertaining to what others MUST do everyday.
I'd like to use Audition to record simultaneous tracks from different sources (microphones). I'm using several USB sound card dongles (each with a single 3.5mm stereo microphone input and headphone/speaker output) as quick and convenient method of creating an array of microphone inputs. Each dongle appears as a discrete audio device with a corresponding input and output.
Fine so far.
Where I'm struggling is in trying to map each microphone to discrete mixer channel. I choose Edit->Preferences->Audio Channel Mapping, but it seems that only channels from any one particular sound device may be mapped to mixer channels at any time. If I assign the microphone from sound card A to Track 1, I can't assing the microphone from sound card B to Track 2. When I try, it alters the hardware device selection and negates any prior channels I've mapped that aren't part of the same device. I hope I've explained the situation clearly.
I'd like to map the microphone from sound device (dongle) A to Track 1, the microphone from device B to Track 2, C to 3 and so forth. But I cant' seem to figure out how to make Audition simultaneously recognize more than any one sound device at any particular time.
Is there a way to do what I'm trying to achieve? Does Audition not support simultaneous mixing/recording from multiple hardware devices? If not, do other products exist that DO support the configuration I'm trying to create which I can use as a front-end - to later mix-down in Audition?
I'd greatly appreciate any helpful insight, guidance, suggestions or recommendations.
Thanks!

remford wrote:
I'm not having any SYNC problems that I'm aware of...
Of course you're not - that's because the system prevents you having them by limiting the physical sources you can use!
Here is a brief explanation of the sync problem - I hope it makes sense:
When an input device digitises a signal, it has to provide a clock source for its A-D device. Normally this clock is locally derived, and the device outputs a digital stream at that rate. But if you have two devices (let's say that they were two external USB ones) both trying to be the input device, then their clocks won't be synchronised - inevitably. Audition, and all other software come to that, can only record at one rate, and if another device provides an input that isn't exactly at the same rate and phase as the one providing the master clock, then fairly soon, you get to a point where the data being inputted won't coincide with the main clock any more, and a bloody great click occurs. On older systems where you could use multiple devices, some allowed for the clock sources to be externally synchronised, and a lot of pro gear still does - it helps a lot when you are using digital mixers and external digital preamps if you can feed everything from a master clock, rather than daisy-chaining it.
When it comes to ASIO, Steinberg made the relatively easy decision not even to try to implement it for multiple devices. Instead, they suggest that you use a single device which is capable of supporting as many inputs as your PC can manage. As an example of how this works, my MOTU device that I mentioned above has eight analogue inputs, and also an 8-channel ADAT input. So, without too much trouble I can connect an external 8-channel digital preamp to it and have 16 inputs in total. It uses an optical connection for the ADAT, but to use it I either have to provide sync from the MOTU to the preamp, or use the preamp as the sync source for both itself and the MOTU, replacing the internal clock. With the only digital preamp I have it's this latter option that makes sense, as there isn't an easy way to get sync from the MOTU back to the preamp. (I should perhaps explain that the preamp has got a sync input, but the only sync output from the MOTU comes from its ADAT output, and that isn't easy to feed back in).
Yes, until you get used to it the concept of syncing devices takes a bit of getting used to - Steinberg's idea was to keep it as simple as possible, even though on the face of it, it's rather frustrating.

[SOLVED] Whirring noise in recording via mic - will HW mixing help?

I'm getting whirring noise in mic recording (adjustng mic and mic-boost does not help) it also happens on ubuntu 11.04; so I guess it is a hardware issue. Strangely, I hear no noise in Windows XP in recording (sound recorder). Whereas in Archlinux, skype, audacity, arecord, gnome-sound-recorder all are having whirring noise.
I guess, it might have something to do with Intel Audio, so I'm planning to buy Creative Sound Blaster Live 5.1 (emu10k1), which has Hardware Mixing feature supported by Alsa.
Do you know any way to "suppress noise" and "echo cancellation" in arch for microphone recording?
Please advise.
We can hear the noise (unamplified/no mic boost) test-mic.wav recorded
arecord -f cd -d 5 test-mic.wav
http://www.mediafire.com/?e6n9t2sr6vwpv73
lspci wrote:00:00.0 Host bridge: Intel Corporation 82G33/G31/P35/P31 Express DRAM Controller (rev 10)
00:01.0 PCI bridge: Intel Corporation 82G33/G31/P35/P31 Express PCI Express Root Port (rev 10)
00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01)
00:1c.0 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 1 (rev 01)
00:1c.1 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 2 (rev 01)
00:1d.0 USB Controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #1 (rev 01)
00:1d.1 USB Controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #2 (rev 01)
00:1d.2 USB Controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #3 (rev 01)
00:1d.3 USB Controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #4 (rev 01)
00:1d.7 USB Controller: Intel Corporation N10/ICH 7 Family USB2 EHCI Controller (rev 01)
00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1)
00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01)
00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01)
00:1f.2 IDE interface: Intel Corporation N10/ICH7 Family SATA IDE Controller (rev 01)
00:1f.3 SMBus: Intel Corporation N10/ICH 7 Family SMBus Controller (rev 01)
01:00.0 VGA compatible controller: nVidia Corporation G92 [GeForce 9800 GT] (rev a2)
03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01)
Last edited by fast_rizwaan (2011-09-08 00:29:10)

hi archers,
I went to the computre hardware market today to buy 'Creative Sound Blaster Live 5.1 with emu10k1 chipset'.
I thought that having hardware mixing (HWMIX) will fix noise, as the sound card is shielded from interference.
But, unfortunately or fortunately, the SBS 5.1 is not available in the market. only SBS 5.1VX model which has a different chipset is available.
So, I retured a bit disappointed thinking that the evil microsoft has caused the soundcard manufactureres to stop support to non MS OS, oh my poor linux hardware!
Based on the following, http://www.voxforge.org/home/docs/faq/f … pling-rate
I tried fiddling with sample rate, and you know what, the whirring noise is gone. It's like Windows (without fan noise suppression and echo cancellation but no static noise)
HURRAY I FOUND THE SOLUTION: setting the sound-card sampling-rate to 96000 (for my intel sound card) fixes the noise!
Here's HOWTO fix microphone noise problem in linux which was bugging me for many years!
1. Determine soundcards in the system
$ arecord --list-devices
output wrote:**** List of CAPTURE Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 2: ALC888 Analog [ALC888 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
soudcar is hw:0,0
2. Determine Sound card's sampling-rate of the sound card
arecord -f dat -r 60000 -D hw:0,0 -d 5 test.wav
output wrote:"Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 60000 Hz, Stereo
Warning: rate is not accurate (requested = 60000Hz, got = 96000Hz)
please, try the plug plugin
Now that we got the max-sampling-rate to 96000, we should set this in pulseaudio's configuration file /etc/pulse/daemon.conf. So first test it:
Just a quick test
arecord -f dat -r 96000 -D hw:0,0 -d 5 test.wav
And surprise, surprise, no whirring static noise in the mic recording in linux. Hurray again!
3. Setting the soundcard's sampling rate into pulse audio configuration
the default sample-rate in pulseaudio is
cat /etc/pulse/daemon.conf|grep sample-rate
output wrote:; default-sample-rate = 44100
we got 44100 and which is disabled
Let's set that to our hardware specific settings:
su -c "sed 's/; default-sample-rate = 44100/default-sample-rate = 96000/g' -i /etc/pulse/daemon.conf"
let's verify the changes to deamon.conf
cat /etc/pulse/daemon.conf|grep sample-rate
output wrote:default-sample-rate = 96000
It's done
4. Restart pulseaudio to apply the new settings
pulseaudio --kill
pulseaudio --start
That's it, now Skype, and all microphone related application will have neat audio! Thanks Archers. hope it helps.
Is there a way to suppress fan noise in pulseaudio, cause the fan noise is bothering me now at forvo.com
I also updated the pulseaudio wiki https://wiki.archlinux.org/index.php/Pu … _Recording
Last edited by fast_rizwaan (2011-09-08 00:27:53)

MIXING help needed!

I have been receiving conflicting advice from people outside of this forum. One tells me to put 0 compression on everything in order to give a boost. Another person says no compression at all, to leave it to the Master Engineer.
I am just concerned if i export the song at a low volume. or is that something the master engineer can adjust regardless of my export volume.
and all the songs on LP should be at the approximate same volume at the mixing stage as well, right?
anyone know of any good advanced books on mixing in GB, or on GB in general?
Thanks!

Hi,
This has to do with compression and limiting.
A compressor decreases the dynamics of audio. So, If you compress audio, the loudest parts become a bit softer and the softer parts become a bit louder. This way, you can make drums, bass, guitar and other percussive instruments appear to be much louder.
A limiter is a little different: it just limits the maximum level of the audio, by quickly adjusting the level.
At the end of the "chain", the Master output, I often use a limiter, a multiband compressor, or a combination. A multiband compressor is 3 or more compressors in one, where each compressor takes a part of the audio spectrum (bass, mid, treble).
If you click-hold the "Setting" button at the top of the Master Output Channel Strip, you can choose al kinds of Mastering presets, that can help you with experimenting. By the way: don't overdo it, or you'll get too much "pumping", which will sound unnatural.
For more backgound info on compression, see http://en.wikipedia.org/wiki/Audiolevelcompression
Kind regards,
Jaap

Compositing - Channel Mixer Help

I've been using AE now for a few years, (since v5.5) and mostly use it for motion graphics. Using CS3 now. But right now, I'm beginning to work through Steve Wright's book, <a href="http://www.amazon.com/Digital-Compositing-Film-Video-Second/dp/024080760X">Digital Compositing for Film and Video</a> to focus more on true compositing.
The author's aim is to keep the lessons application-independent and present concepts in a way that would be applicable to most software packages. (However, considering the author's frequent references to "nodes" and his well-known status as a Shake user & <a href="http://movielibrary.lynda.com/html/modPage.asp?ID=408">Shake instructor</a>, it's apparent that using Shake, while reading this book, would lead to the least bumps in the road, when attempting to follow along.)
But I'm not using Shake, I'm using AE and have encountered a small bump in the road. Regarding luma keys he mentions how luma values are represented in the individual colors R,G,B. To cut to the chase, he says, "One standard equation that mixes the right proportions of RGB values to create a luminance image on a color monitor is:
Luminance = 0.30R + 0.59G + 0.11B"
Great. But I can't figure exactly how to mix the RGB values with the above proportions in AE. It's quite easy & straight-forward in Photoshop, using the Channel Mixer. <a href="http://www.geocities.com/kevincbrock/channel_mixer.png">Example here</a>. But AE's Channel Mixer does not seem to work quite the same way as it's counterpart in PS. There are values there, for instance, "Red-Red", but they seem to run in values from -200 to 200, not percentages, like PS. So my question is, how do I achieve the same channel mixing results in AE that I can in PS?
I'll add this: I was able to achieve the desired result in AE, but by using a round-a-bout sort of tactic. Instead of using the "Channel Mixer", I applied the "Set Channels" effect to my sample shot. By changing the "Set Red to Source 1's LUMINANCE", then doing the same for Green and Blue, setting them all to LUMINANCE, the resulting image is identical to the sample shot that was tweaked in Photoshop. Again, it worked, but I'd like to know how to change the proportions of RGB to other values than those represented in the formula above.
Hope this all makes sense.
Oh, and while I'm here.....I'll throw this in: In AE, when you apply the "Curves" effect, can you input precise values for input / output. Again, this is simple in Photoshop, but unless I don't have some display items turned on in the Curves editor in AE, I don't see a way to enter precise values. Is this possible?
Thanks in advance for any input.
EDIT: for some reason, the links I've posted aren't showing up properly. So if your'e interested in following any of them you'll have to copy/paste on your own. Don't know why......

>Great. But I can't figure exactly how to mix the RGB values with the
>above proportions in AE.
You could use expressions... You create sliders for each input, then feed their output to the actual effects. Something like
//----begin expression
linear(value,0,100,-200,200)
//----end expression
should do in most cases. Still, be careful. Not everything written in that book directly applies to AE. It does not figure in any specifics of color management nor different Gammas which could massively skew results because the source footage will already be pre-adjusted and the comp will add another level of adjustments.
>Oh, and while I'm here.....I'll throw this in: In AE, when you apply >the "Curves" effect, can you input precise values for input / output.
Nope. AE's Curve effect will adapt to the project bit depth as a lot of other effects, presenting you with the problem of which values to actually use. This would only make sense if the effect itself would be independent from AE and process in float all the time and only the inputs were presented as different normalized ranges. There was/ should have been an alternative Curves effect from Frischluft, but at this point development has reached a dead stop, so it's doubtful it will ever be unleashed upon the world.
>Don't know why......
HTML-Code functionality had to be disabled thanks to spammers. :-|
Mylenium

Linking L9 to hybrid Yamaha digital mixer - help!

Hi
I recently bought one of the hybrid Yamaha n12 mixers mainly to enjoy remote hardware mixing on Logic. I have really struggled to get the MacBook Pro talking to the n12. I have downloaded and installed all the manual says I should from Yamaha's update page for the n12 yet (a) cannot get the mLAN audio download to install (it keeps failing) and (b) neither can I get the Yamaha Steinberg FW Driver to appear on Audio MIDI Setup as it should.
Has anyone else out there has had similar issues?

What operating system are you using?
Are you using these drivers:
http://download.yamaha.com/search/product/?language=en&site=usa.yamaha.com&categ ory_id=16255&product_id=592444
Use the drop down to filter for your OS.
Is mLan included in the Firewire driver + Tools?
Or are you using the separate mLan download... that will never work unless you're using very old OS software, also, have you done the firmware updates?

Audigy 2 zs - Line-in is missing in the mixer - Help pleas

Hello,
When I switch to recording in the mixer, theres is no Line-in slider.
Have the latest driver installed.

You must select "Analog Mix" in the record slider. That's the line in.

Audio mix help

What is the best way to do an audio mix with multiple channels where you want to change the level of one complete channel as opposed to the individual clips one at a time in that channel?
Thanks Angus

Use the track tool to select the one track and Modify>Levels to change the levels of the track.

Itunes 9 Genius mix(help)

Can you import your mix playlist to your ipod? if so how?

I have an iPhone 3G and an iPod Classic 120GB. The iPhone "Music" tab shows all my Genius Mixes. My iPod "Music" tab does not show them. If I'm thinking of them not as a 'static playlist,' then how is it syncing them to my iPhone, and telling me that they're too big to all fit? It must have, at the time it's synced, a static size and set of songs. And if that's the case, it should also work on my iPod Classic, since it's just a list of songs (that you're not allowed to see, for some reason).

SoundBlaster mixer support under plain DOS?

I have the following problem at hand - I want to run some DOS based programs (mainly old games and demos) which use SoundBlaster.
When the integrated AC97 codec is enabled in BIOS and also the legacy SoundBlaster support (with 220, 5 and 1) and I have set the BLASTER environment variable, then the SoundBlaster is detected successfully, but the sound remains muted - it works, but can only barely be heard when I turn the external amplifier up extremely high. I believe the problem is in AC97 codec and BIOS which does not set the registers for the AC97 chip properly.
Does anyone know of a DOS based AC97 mixer controller or any info on how to turn off the mute and turn up the volume of AC97 codec _in DOS_ so that DOS based programs can use the SoundBlaster emulation?
My rig consists of MSI K7T Pro2-A under a 1GHz Athlon TBird and 512 MB of memory.
Bratac

It seems that I got this thing worked out. From the link given by Bas (thanks!) above I got the the AC97 mixer by Alex Mina, which actually does the trick, though from under Windows.
So, I installed a barebone Win98SE, but didn't install VIA audio driver. Windows detected the BIOS enabled SoundBlaster Pro emulation and installed the crude SB Pro driver.
At first it was like before - max amplification revealed sound working, but being muted like before. Now Alex's AC97 mixer helped - I was able to un-mute the global volume and PCM line volume of the AC97 codec, and voila - there was sound.
Now I exited to MS-DOS and most DOS things work with SB Pro settings, even those that require the CPU _not_ to be in virtual x86 mode !

I did my first monitor analysis: Now what?

Hello everyone,
I'm a student film music composer and still very new to producing music, especially mixing. I'm using my fathers Alesis M1 passives and a QSC USA400 power amp in his studio; a converted garage thats been sound proofed into a very dead room (for pictures see: http://www.shopjt3.com/studiopicts.html). I've been generally unhappy with the way my music translates to other systems/environments, and am unsure how to fix this problem. Is it my monitors? Is it the room? Is it user error? I just dont know! For example, under the advice of a friend, I recently added my cheep home stereo subwoofer to my monitor chain, and was AMAZED at how much better my mixes became! I was no longer flying blind with low end frequencies, which in turn made my whole mixes less muddy! Seeing first hand the improvement this made, I'm wondering if the same thing will happen if I upgrade my monitors to a more "professional" set.
Hoping to gain more insight into my room/sound, I purchased the Alan Parsons monitor analysis package (pink noise test tones with DB monitor) and came up with some interesting results which I graphed here: http://www.shopjt3.com/images/monitor_analysis.jpg Note, there are 3 graphs: Alesis LRSub; Alesis L+R with no sub; and Sub only. My Monitors are 20 inches from walls. Distance between monitors is about 47 inches. Distance from monitor to my working position is about 27 inches.
I have no clue where to go from here. Are the dramatic changes in frequency information due to the monitors, the power amp, or my room? Should I add an EQ to adjust for this when mixing, and then take it off on the burn? Should I use this EQ to make my monitors as flat as possible? Would I be better off ditching the whole system for something like the Adams A7's + a Blue sky sub? I really have no clue what to do...why dont they teach this stuff in school?!
I write music for films as well as progressive rock. My goal is to get my stuff to translate as well as possible to other home stereos / TVs & movie theaters. I hope to eventually hire out my mixing / mastering, but dont have the budget to do so yet. Of course, the catch 22 is your music needs to sound professional in order to get the professional gigs to afford professional mixing help! Here is an example of my most recent film score: http://www.fieldsfaraway.com/final.html (some of you have already heard this). Does this sound professional? How well does it translate on your system? I'm working on another project now, and will post clips for comments soon.
Sorry for the long post, but I'm new to this stuff and really have no clue what to do. However, I am determined and willing to learn, and would appreciate any comments, questions, advice you're willing to share. Sorry this post isnt directly Logic related, but the best information I've found on the net has come from these forums.
Thank you in advance for your help and reply!
-Jonathan

Jonathan Timpe wrote:
I just remember using $99 audio-technica headphones at my dorm to mix, and ALWAYS having my mixes turn out extremely muddy and dead sounding (because I couldn't hear any bass I would boost it up thus muddying my mix). Perhaps a quality headphone would do better? Would headphones be my main monitors, or just to check my mix?
$99 headphones are unlikely to be suitable. Decent (read: expensive!), open-backed cans are the way to go. Find a store that lets you try out the headphones - indeed take your old audio-technicas with your to compare - I think you'll be surprised at the difference.
any other suggestions? I've recently read that should never be perpendicular to a wall (which is where mine are now). Is this true? I've also read its best to have them as far away from wall as possible, yet I dont want my desk to be in the middle of the room (plus I've NEVER seen this in any studio pictures I've ever seen). But the bottom line is the sound, so I'm willing to do whats necessary.
That stuff you've read is not wrong but, as usual for this subject, it always "depends"
What's behind your speakers and how far they are away from the rear surface affects the sound considerably. Bass pretty much radiates in all directions - including to the rear, and it will refelect from the rear wall and back to you. What you hear is the direct sound and the reflected sound combined. This can result in a reduction of bass or an increase. If the distance is big enough your ears/brain are capable of identifying the different componants - or the reflection may be quiet enough not to matter if the distance is very big! In most real rooms this distance will be small and you hear the combined effect. The other walls affect things in the same way but the closer the wall the more serious the effect can be. Having said this the design of many speakers take this rear wall issue into account. There is often advice on speaker placement from the manufacturer for this reason. My Mackie speakers have a switch to set their location to "corner", "near wall" and "far from wall" - its essentially a bass lift eq (yes I know this kind contradicts the EQ advice - lets not go there for now .)
Often you have to experiement to find the best position for the speakers and it entirely depends on the model/make.
In general, with acoustics, anything symetrically shaped is not good. A rectangular room is better than a square for eg and even better is a room with non-parallel walls (like many top studios). Placing speakers non-symmetrically, such as one closer to the corner than the other can help sometimes as it makes the reflection patterns of sound more random. More random is a good thing. Having said this you can end up with one side sounding more bass-heavy than the other for eg and that is not desirable. Guess what?... experiment!
As a start point I'd suggest placing the speakers about 1 to 2 feet from the wall (I'm being realistic in what you can do in a garage, and I'm guessing what your garage is like!). Sit infront of them so that there is fair amount of space behind you but don't sit at the centre of the room - its better to be seated somewhere between the speakers and the centre of room too, rather than the centre of the room inbetween you and the speakers (if you follow me!). Now set the distance between the speakers so that there is roughly the same distance betwen them as there is to your head - so there is a equilateral triangle between your head and each speaker. Play some music, listen and then move yourself and the speakers around a bit and see what difference it makes.
Oh and its usually best to place your speakers on the longest side of a rectangular room, when there is a choice.
As a final comment on mix technique. Never forget that at the first level you are trying to make instruments sound real. There is a tendancy to try and make the instruments sound bass heavy and "full" or "shiny" with extra top-end. Firstly look for truth. Imagine what a real cello sounds like and make your recorded cello sound like this. Indeed its worth trying to mix without any eq or compression to start with. It can surprise folk how well a mix can sound this way. Punch and vibrancy can be added at the mastering stage too.
Indeed when monitoring is less than perfect always be cautious with the eq. The less you tweak the less you can mess things up!
Thanks again for your wonderful post!
I always write long posts when I'm trying to avoide work:-)

Korg MicroKontrol with Logic Pro 7.1 & 7.2?

Has anyone actually got the Korg MicroKontrol to work with its supplied Logic scenes & If so how do you do it?
7.2 is suppose to fix the 7.1 problem (as below) but it don't!
More Control Surface Support: The M-Audio iControl to the Frontier TranzPort, Tascam FW-1082 and US-2400, JLCooper boards, and even the Korg MicroKONTROL now works as a control surface in Logic 7.2 A new display in the mixer helps you keep track of what you’re controlling.

Has anyone actually got the Korg MicroKontrol to
work with its supplied Logic scenes & If so how do you do it?
The mK either works in Native mode (which is what the control surface support docs mean) and in this mode, its internal scenes are ignored.
Or you can use the regular mode, but the Logic scenes aren't terribly good.
I'm using the mK in 7.1 in both modes without problems, so what exactly is the problem you are having?

Recommend Workflow for 2 writers to share RH

I am currently working on a large (modular) help system
consisting of 10 projects. I have been told that RH 7 is not suited
for working across networks. But the other writer and I may have to
work on this mixed help system. It currently sits on my C: drive.
Can someone suggest the workflow or an article that points out
where to place the files on a network, how to access them
(RoboSource Control ?) and the gotchas one can expect working in
this environment (with two people over a network).
We are using RH 7 (WinXP) in a network. We currently create
and store our own projects on our C: drives, but may have to both
work on the big modular help system in the near future.
Thanks in advance.

Hi forme3d -
There are essentially three choices:
1. Take turns working in the project, passing the fileset
back and forth.
2. Use some form of source control.
3. Split the project up using the merge feature.
Option 1 has limited usefulness, so you should take a look at
both Option 2 and Option 3. There is some good information in these
forums (fora
) about merging. Do check out the extra information
provided by the resident experts on their own sites.
We use RSC successfully, but not without considerable pain,
as well. Our projects are very large and complex, and sometimes
things just break down. Overall, I've felt like it's been less
painful to use RSC than to not use it. I don't get agreement on
that from every writer, every day, though! What I'm trying to say
here is that I wouldn't recommend using RSC just for the fun of it.
Our decision driver is whether we need to have multiple authors
working in the project at the same time. If the answer is yes, then
we set up the project in RSC. If the answer is no, then the writers
are relieved.
G

Looking for help on how to start mixing ?

I have a two songs done and Iam starting to get how to record but thats all I know . I need help moving a head on mixing and putting it on CD Thanks

Take time to learn as much as you can about dynamic processing - especially compression. Also, listen to a few of your favorite CD's through your monitor speakers to get an idea how you may want your mixes to sound.
Take advantage of panning your instrumentation so that you have a wide mix as opposed to everything sounding straight in the middle.
Have fun!!

Mixer help.

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