Slippers

HoWdY!!

I have a SB0220 Live! card, using the kX3538M version of the driver, and using those ProFX plugins.
I believe that there are stickies posted that have links to these newer versions of the driver and plugins.
I am using Cakewalk Sonar Producer Edition 6 for my musical recording, arranging and editing.

This is what I am currently using as my DSP for recording ...

http://www.fileden.com/files/2006/7/...Record_DSP.JPG

I have the audio playing from FXBus(1), the MIDI playing from FXBus(2) and my Behringer MXB1002 mixer into the Line In port on the SB0220.
I don't know if I need to have the rest of the FXBuses attached ... I don't know what I need them for.
They don't do anything because I can only record one track at a time from the line in.
As you can see from the pic below, I am active and ready to record.

http://www.fileden.com/files/2006/7/...cord_DSP_2.JPG

OK ... I take each midi track, mute and archive the rest (not heard, not processed) to take the load off the CPU, then using KxASIO Right In02 and KxASIO Out02 I record the SoundFont output to a new Audio Track.

This is what Each Track sounds like mixed down into a single stereo track.

http://www.fileden.com/files/2006/7/...CS/Mixdown.mp3


I put a compressor plugin between the LineIn and the mixer to keep the peaks in control, and then added some everb and delay echo afterwards. I don't know if this is right or not, any other opinions, ideas or hints welcome.

Ok ... I'll leave this here right now until I get some feedback.

Thankx,
Dennis

Russ

You should not need the other FxBus' for what you are doing (from what I can tell anyway)

I am not sure if you meant to do this (I assume that you did), but you are only taking the right channel from Line In.

You are not recording MIDI, so I assume that you are just playing along with it? (I think you mentioned something about that in another thread, but I do not remember all the details).

You have the MAIN output enabled for Line In, in MX6, so I assume that this means that you want to monitor Line In pre-recording? Note that it seems that you are also monitoring Line In through the ADC plugin as well (Line In is enabled on the right side of ADC, which is usually used for monitoring via the AC97 codec (i.e. dry signal / no DSP effects) or for Stereo/Mono Mix recording). You might want to disable that so that you are only monitoring through MX6.

I assume that you are using FxBus (0/1) to hear the playback of your recording, or the mix-down, etc?

As for what is right, that depends on what you want to do... From your config, you have Reverb and Delay applied only to the recording outputs of MX6, so those effects will be applied to Line In and recorded, but if you are monitoring pre-recording, you will not hear those effects while monitoring, etc. Also it might be better to use Reverb (and possibly Delay as well) as a send type effect, so you can get a WET/DRY mix instead of straight Reverb (you could use a Stereo Mix plugin for this, or use the MX6 sends).

BTW: Nice pics, but you do not necessarily need to label all the outputs for the plugins, so long as we know how the plugin is configured (unless of course you prefer to do it).

Slippers

HoWdY!! Thankx for your reply Russ!

Great .. I was thinking that too. I can just leave them disabled then or totally delete them ... can't I?

Yes. I found that I was getting a much stronger right side signal on the LineIn from my Behringer mixer so rather than fiddle with the levels for the left and right sides, which was a mono input anyways, I thought that I could feed both sides of the MX6 from the stronger input on the right side.
Is this a good or bad idea?

Well, I WOULD like to record midi, as well as voice at the same time, but not always. I didn't really work on the midi recording much as I was more interested in the audio recording first.

I don't understand your words.
Could you draw me a picture please?
It's just the way my brain is wired. I'm a carpenter and have to see a picture to understand. If I can't see it in my mind's eye, I can't build it.

I don't want to monitor pre-effect but post effect signals.
Don't I have to have those lines from those contacts going somewhere? Do they work if they aren't attached to something?
Would that be why I get that low screaming humm sometimes?


Yes .. that has to be connected to MX6 to hear ANYTHING ... right?
The sounds won't play oitherwise? Right?

Ok .. that's a boo-boo then. I wanted to HEAR what was being recorded ... not just the pre-effect Line In.
What do you mean "Send Type Effect" and where would I attach that?
Isn't it better to have less stuff in your DSP than more?

I just want it to be able to simply record my midi and sound, hopefully at the same time, play back nice, have a couple of basic effects to improve things a bit(compression, Delay, reverb), and be able to mixdown to an MP3 file of reasonably good quality stereo that I can burn on a CD and send to my mum. (grin)

Could you possibly take this DSP I have made and show me what was wrong and what would improve it knowing what I am trying to do please?

Ok ... I did that because I was confused why the inputs on the ASIO5.1 went ASIO(0), ASIO(1), ASIO(2), ASIO(5), ....
What happened to ASIO(3) and ASIO(4) ??
What math do I have to do to find out where they went?
I wanted to make sure that there was no errors and that anyone responding to the message would be able to do so accurately.

I will omit those lables the next time I post something.

OK ... I feel like this is getting me outta the fog of confusion a little and I am getting a better understanding of how this entity works.
Thank you very much for your time and efforts on my behalf.
I sincerely appreciate them!

Thankx
Dennis

Russ

OK let me tackle this a little at a time, as some of it is a little hard to explain.

Yes, you can unload the plugins for the unused FxBus channels.

That is fine if it is a mono signal anyway, but I wonder why you are getting a stronger signal on the right channel...

That is fine, and can be useful for plugins that are more model specific (i.e. ASIO 5.1), but for other plugins it is not necessary (i.e. with MX6 and kxlt, the input/output names are not important (they are bacially the same for everyone), what is important with those plugins is to be able to see how they are connected and configured).

Some effects have there own WET/DRY mix controls while others do not.

There are (mainly) a couple different ways to use effects.

As an insert type effect, where there is only one signal pathway, and 100% of the signal is sent through the effect(s), and the signal is taken from the end of the effect chain to be recorded, etc (like guitar effect stomp boxes).

The other way is to use an effect send type of setup, were you split the signal and send a portion (whatever amount you want) of the signal to the effect, and a portion of the signal bypasses the effect, and then the two signals are recombined (mixed together) at another point beyond the effects output, so that you can control the WET/DRY mix (and thus the amount of effect that is applied to the signal). MX6 has two send outputs that can be used for this. i.e. Connect the effect(s) to the send outputs of MX6, and then adjust the SEND1/SEND2 sliders in the MX6 GUI to control how much of the signal (for each input) is sent to each send, Then you can connect the output of the effect(s) to an unused input of MX6, and thus mix it in with the dry signal (by enabling that input for Main and/or Recording and adjusting the levels however you like).

More to come...

Russ

Ok, now onto ADC/AC97. This is hard to explain as you need to now a little about the AC97 codec (which can be a little different for different card models), but I will try to explain.

For most card models, it works something like this (this is all inside the AC97 codec):

The AC97 page of kxmixer, as well as ADC, control the AC97 codec.

You can enable any AC97 source(s) to be sent to the AC97 Mixer (parrt of the AC97 codec). This is what the sliders and mute buttons at the bottom of the AC97 page of kxmixer, and the sliders and enable buttons on the right side of the ProFx:ADC plugin do. On most card models, the AC97 mixer is fed to the analog front speakers, so anything sent to the AC97 mixer will be heard in the front speakers (i.e. monitoring via the AC97 codec itself). Many drivers actually use this as the playback slider(s) in the Windows Mixer (which are separate from the recording section of Windows Mixer (i.e. you can record with the playback slider muted)). So if you are familiar with other drivers, you can sort of think of those sliders and buttons as the playback section (for AC97 sources) of the Windows Mixer.

You can also choose a Record source within the AC97 codec. This is the AC97 source selection combo-box that you see on the AC97 page of kxmixer, and the combo-boxes that you see on the left side of the ADC plugin. The source can be any AC97 source (Line In, Mic, etc), or the AC97 mixer itself (when the source is set to Stereo Mix or Mono Mix, the AC97 mixer is the source). Again, with other drivers this would be the Record section of Windows mixer (where you select a source, and can adjust its volume (except Windows Mixer usually has separate volume controls for each source, but it works the same way (controls the Recording Gain)).

So, the record source that is selected is separate from the AC97 Mixer, unless the AC97 mixer is chosen as the record source (in which case anything sent to the AC97 mixer will be recorded).

So, except for the case where the AC97 Mixer is the record source, the left and right halves of the ProFx:ADC window are separate (used for different purposes). One side is for recording, and the other for playback/monitoring.

The Mic boost option boosts the MIC, so it applies to both sides.

The pins on the ADC plugin are the AC97 record source (signal from) into the DSP, so that you can apply DSP effects to the signal before recording. You cannot do this on the playback/monitoring side (when monitroing via AC97) as it is direct to the speakers (thus you will not hear DSP effects when monitoring this way).

Here is a picture of ADC for reference.

Again, this is a bit hard to explain... Hopefully I did not confuse you more.
This is how it works with most card models, although there does seem to be some models that do it a little differentley (I will not discuss those models as I am not completely sure of the inner workings of the codecs used on those models). I am prety sure that your model works like this, but you can test to be sure:

Save your config so that you can reload it whenever you want.
Clear the DSP.
Load ProFX:ADC and kxlt only.
Do not connect them (so there is no direct connection from ADC to the speakers in the DSP).

Lower the Line In volume slider (so it is not too loud) on the right side of ADC and enable Line In (on the right side), Play some audio though Line in and most likely you will hear it (adjust the volume slider as necessary) in the front speakers (even though there is no DSP connection from ADC to the speakers). Note that it does not matter which source is selected as the record source (on the left), you can still hear Line In and adjust its volume (i.e. monitor it via the AC97 codec).

Alternatively, if you disable the Line In slider on the right, select Line In as the Record source on the left, and connect ADC to the WinMM Recording connections of kxlt, you are able to record from Line In using Sound Recorder, etc (although you will not be able to hear it).

i.e. they are seperate in most cases...

The point is to show that you can monitor the signal via the AC97 codec, and if your intention is to monitor the signal via another means (via the DSP before recording or after recording), then you should disable that (otherwise you are hearing the signal at different points at the same time which is usually not what you would want to do (unless maybe you want to monitor the dry signal using headphones, and monitor it another way from one of the other speaker connections too)).

Does that make sense to you?
(Are you sorry that you asked? )

More to come (when I have a chance)...

Slippers

HoWdY!! OK ... great stuff Russ!
I seem to be falteringly following so far. An illustration or three would be helpful (please!!)
Could you take us along the path that the signal follows and show us what you think is an efficient way to accomplish the task I am trying to do ... for instance ...
I'm still stumbling on the connector thingies ... maybe I'm bogged down with the "wire connections" mind's eye image I have.
I'll go and re-read this a few more times .. with my DSP open as reference (wink)

Really great insightful discussion Russ! Please enlighten us further with your knowledge and experiences!

I can't wait to read your further postings!

Blessing,
Dennis

Russ

Did you want to record MIDI (soundfont output) to the same track (i.e. mixed in the DSP) or to a seperate track?

Did you want to monitor in the DSP (i.e. pre-recording) or from your application software? (monitoring, i.e. not playback of the recording/mixdown)

Slippers

I think I would like it to go to a separate track so that I can set the levels and stereo separation of each individual instrument before mix down to Master MP3 (192Khz) file.

I think I would like to have the Cakewalk software keep track of the monitoring.
I have an older mixer unit called a "Peavey StudioMix" that has servos and can control a bunch of stuff to and from Cakewalk. It was a partnership with Peavey and Cakewalk and this is a really handy tool to be able to use with channel automation, levels, jogging wheel, etc. and such.

I would also like to see your progression (Figure 1, Figure 2, Figure 3, etc.) as you build this DSP so that I can understand WHY you arrange things the way you do them.
A short blurb on what each step does would also be excellent!

Again, it's really great that you would take the time and energy to explain this and help out mywell, as well as probably a whole bunch other people who are reading but not posting.

Ok folks ... if you are reading this and Russ' explanations and such are helpful, even if you just post "Thanks!" and your handle, I think it would show respect and appreciation for his efforts, knowledge and experience!

Say thankx for helping!

Thankx,
Dennis

Slippers

[quote=Russ;1070919]

[SNIP]

Basic Example One:

Russ Explain Figure 1

http://www.fileden.com/files/2006/7/...lain_Fig_1.jpg

Russ Explain Figure 2

http://www.fileden.com/files/2006/7/...lain_Fig_2.jpg


Russ Explain Figure 3

http://www.fileden.com/files/2006/7/...lain_Fig_3.jpg


OK .. This is starting to make sense!

Thankx Russ!
Dennis

Slippers

HoWdY!! Ok Russ ... using your explanation as a base, let's say I want to record an audio track from a song playing in a website song player.
I am seeing all the action on FXBus(1) ... how do I grab that to a track in Cakewalk and record the sounds that I am hearing?

Would this DSP work?

Russ Explain Figure 4

http://www.fileden.com/files/2006/7/...lain_Fig_4.jpg

Russ

No that DSP would not work.

ADC is for AC97 sources mainly (hardware inputs on the card), so generally you would not use it for Wave audio (you could but it is more complicated). The easiest setup to record audio from FxBus (0/1), would be to take the signal directly from FxBus (0/1).

i.e.
Replace ProFX:ADC in your config (Figure 4) with ProFx:SRC set to FxBus (0/1). (Sorry no pic, as it really should not be needed here).

BTW: In Figure 2 and Figure 3 (from your other post), you did not disable the Line In slider on the right (i.e. the little blue indicator (enable/disable) button is still lit up).

Russ

The way I arrange my DSP is really pretty basic.
Load all audio source plugins that I want to use first.
Add any effects or other plugins in the order that I plan to connect them.
Add any output plugins last.
(possibly add peak plugins at various points to check signal levels as the last step).

I am not sure that I want to abuse the free image hosting services too much, to show intermediary steps that can just as easily be explained with words. Also, since I start with a clear DSP, you can easily see the order that I load the plugins by the number (in square brackets) assigned to the plugins in the DSP window.

For what you want (as I understand it):

You want to monitor from the host software. I would use ASIO (0/1) for this, which means a ProFX:SRC plugin set to FxBus (0/1) is needed. Since FxBus (0/1) is also the default for Wave audio output, it can also be used (to hear) for playback of the mixdown, etc.

You want to record MIDI soundfont output (kX Synth output), which means a ProFx:SRC plugin set to FxBus (2/3) is also needed.

You want to record from Line In, which means the ProFx:ADC plugin, with Line In selected as the record source, is needed.

That takes care of the audio sources. I would load them first (the order does not matter).

Now for the effects:
From your first picture in this thread, the APS Compressor is the first effect (it is the closest in the chain to an audio source), so that is the first effect I would load.

You want to use the APS Reverb which has no Wet/Dry mix controls, so I would use this in a send type of setup. For this I need a plugin to control the amount of signal that I want to send to the effect. I would use a simple St Vol plugin for this (this is the next plugin I would load). Then I would load the APS Reverb plugin itself. Next I need something to mix the dry part of the signal with the part of the signal that had the reverb applied to it. I would use a Stereo Mix plugin for this (this is the next plugin I would load).

You want to use the DelayB plugin which does have Wet/Dry mix controls, so I would probably use this as a simple insert effect (you really need to play around with all the possibilities to figure out how you prefer to use these effects). This is the plugin that I would load next.

You did not mention anything about any effects on the MIDI signal, but if you wanted to add effects there I would load all of them next (in the order that they are to be connected).

Next, since ProFx:kxlt does not have any volume controls, I would want something to be able to control the volume to my speakers. I would probably use a GainHQ plugin for this, so that would be the next plugin I would load.

The outputs:
To get the signal to the speakers we are using ProFx:kxlt, so I would load that next.

To get the signal to the OS (and your host software) to be recorded using ASIO, we need the ProFx:ASIO pluguin, so I would now load that.

That is pretty much it. Add some peak plugins if you want to check the signal levels at any point in the DSP.

Image 1
Image 2

I am using ASIO 4/5 for recording Line In, and ASIO 6/7 for recording MIDI. The pins that I connected to on ProFx:ASIO are the ones that correspond to those ASIO channels (currently) on my card.

I assume that you know how to configure your host software for what you want to do.

This is just one possible config, there are many ways it can be done.

-Russ

Slippers

HoWdY!!

Russ, that answered SOooOooOooOoo many questions and showed me my basic conceptions were incorrect!

I was thinking like I was wiring an electronic circuit. This is a different concept from that but now it makes sense!
Thank you so much!

Your pictures makes things very clear and I understand now!!

WOW!!

Ok .. I gotta go try this now and see if I can get my recordings to work properly!

Blessings to you my Good Man!
Dennis

Russ

It is more like connecting audio components together with virtual audio cables.

Russ

BTW: I left them out of the picture, but the APS Compressor plugin should really have a x4 plugin right before it, and a div4 plugin right after it (it is related to an issue with that particular plugin (mentioned in another thread))

i.e.
ADC -> x4 -> APS Compressor -> div4 -> etc.

Slippers

HoWdY!!
Russ ... I created the DSP exactly like your picture, but it didn't work.

It would play the music for a few bars and then break down into a cacaphony of noise and repeated sounds.

Could it be that I am using an SB0220 and your aren't?

That's why I put all the numbers besides my plugins to show what they had as labels.

When trying to attach to ASIO(4/5) I found I had (0), (1), (2), (5) and up but no ASIO(3) or (4).

In Sonar, in the Options/Audio preferences I am using kX (00) for In and kX(00) for outs. The Output and Recording Timing Master also set the same too.

Where do I go from here?

Bummer dude!
Dennis

Russ

Make sure that your are not using too much delay (repeated sounds (DelayB plugin)), and make sure your latency (ASIO) setting is not too low.

My card is an SB022x card also (SB0228), and has the same ASIO mapping issue (but my mapping may be different from yours). You need to figure out the specific mapping for your card, with the host software you are using, and make the appropriate connections for your situation.

The numbers on the plugin are not really important. What is important is that you do the mapping so that you know which pins (ProFX:ASIO) go to which inputs in your software (again, I cannot do this for you as it may be a different mapping for me (and I do not have the same host software as you)).

I would use kX 04/05 (for stereo recording) for Line In and kX 06/07 (for stereo recording) for MIDI, but again you need to do the mapping to figure which pins to connect to on the ProFx:ASIO plugin.

kX 00/01 (stereo) for monitoring (output) is right.

Russ

Ok, I just tried the Sonar 6 demo so I can see what it looks like (how it labels things, etc).

Here is how I configured it.

Master:
kX ASIO... kX Out 00

Inputs:
Audio 1:
Stereo kX ASIO... kX In 04

Audio 2:
Stereo kX ASIO... kX In 06

This worked fine for me using that DSP config.

Your Sonar config should be similair, but you may need to connect to different pins on the ProFx:ASIO plugin.

Slippers

HoWdY!!

OK ... done ... everything I took down to 50% of where it was origionally default.

Ok .. I think I understand what "mapping"means. It's connecting things together correctly to get it to work properly.
What process should I use to "map" these pins?
I realize it is probably hit and miss ... hunt and peck until you find the right pin, but what is the process used to do just that? How do I know that I've hit the right pin? Is there going to be blasts of screaming feedback when I hit the wrong one?

I think I'm pretty up to speed as far as the other things go. Your explanations here have been very educating. Thank you very much for your kind help.
This mapping thing will probably throw open the floodgates of understanding once I understand how THAT part works.

Thankx,
Dennis

Slippers

HoWdY!!

Ok ... nice program eh?

Ok ... I had mine setup that all the tracks had the same OUT and IN just like what you have above.
Do these all have to be on different kXASIO Ins and Outs?
ie: I have 9 audio tracks ... all of them were using kX ASIO In 04 and kX ASIO Out 00
Do I have to have Audio Track 1 on 04, then Audio track 2 on 06, then Audio track 3 on 08 ..etc .. ??
That actually makes sense ... I think ... no wonder I was getting the WHOLE mix all the time on any channel I tried to record on .. hmmm ... OR am I wrong about this?
Sonar help files are NOTORIOUS for being obtuse and incomplete!


OK ... I'll wait for your reply to clarify this situation.

I am also upgrading my CPU by about 300% in a day or two and I think the more powerful CPU will help things along as well.

Thankx again for your reply!
Dennis

Russ

The mapping issue is that the numbers that you see on the ProFx:ASIO plugin (or epilog) may not match up with the numbers that you see in your host software (and there are 2 missing). Because the numbers do not match up, you have to figure out for yourself what the numbers should be (i.e. pretend ProFx:ASIO is not labeled at all, and it is your job to label it (it is really not as bad as that, but just so you understand what I mean)).

The mapping stays pretty consistent (within the same host software), so you should not have to do this often.

In Sonar, if your arm a track for recording, you should see movement on the level meter (in Sonar) when connected to the right pins for whatever ever ASIO channel that track is set to use (provided you have audio playing).

Also, if monitoring (in Sonar) is set up correctly, if you enable monitoring for a track, you will hear your audio when connected to the correct pins for the ASIO channels that the track is configured to use.

Connect to the pins one pair at at time (i.e. first 2 pins, then 3rd and 4th, then 5th and 6th, etc) until you get a signal in Sonor. As I said before, if the numbers are off, they are usually off by 2, so when you figure out the first one, generally the rest will follow in order (i.e. shifted by the same amount), so it really is not as difficult as you might think.

Every track that is active should use a different input (unless you want to record the same audio to multiple tracks).

i.e.
You connect to 4 pins on ProFx:ASIO:
That is 4 separate mono inputs, or 2 separate stereo inputs.

As for the output, that is up to you. If you wanted to monitor each track on a different set of speakers, you could configure it to use different outputs, but generally I use the same one for all (the DSP config is designed for using just the one (stereo) output).

Slippers

HoWdY!!

OK .. understood. That makes sense now to stuff that didn't before.

BINGO! Just what I needed to see. This now this all fits into place.
I was always wondering what the heck was going on there!

Active .. do you mean ARMED for RECORDING? That's the term Sonar uses I beleive.

So let's say I have written a midi track and want to record this midi onto an audio track using the SB 0220's SoundFonts.

I would mute all the tracks that I don't want to record, and set the DSP to record from the midi input (FXBus 2/3) and then using the corresponding kX IN for the midi signal in Sonar, arm an audio track and just play the midi and it should record that midi audio ouput?

Ok ... using your example above, if I have make 8 SRC inputs for the 8 midi tracks I could one side of each source to I'm guessing the MX8 plugin would be needed ...

FXBus (2/3) --- Sonar kX_IN 02 Left (audio track 11)
FXBus (2/3) --- Sonar kX_IN 02 Right (audio track 12)
FXBus (3/4) --- Sonar kX_IN 04 Left (audio track 13)
FXBus (3/4) --- Sonar kX_IN 04 Right (audio track 14)
FXBus (4/5) --- Sonar kX_IN 06 Left (audio track 15)
FXBus (4/5) --- Sonar kX_IN 06 Right (audio track 16)
FXBus (6/7) --- Sonar kX_IN 08 Left (audio track 17)
FXBus (6/7) --- Sonar kX_IN 08 Right (audio track 18)

... to record 8 mono midi tracks at the same time on 8 audio tracks at the same time?

... OR ...

would I have to, having 8 midi tracks, different instruments, just use the same midi in and audio out, just add new audio tracks and arm them, but just disable the tracks I didn't want to record. This way each audio track has only one instrument on it. The downside is that I have to play the song 8 times to get all the 8 tracks recorded individually.

Next ... what about SYNC between these midi and audio tracks?
In Sonar I can choose to use the midi, internal clock, TMPSE, or sound card clock. Which one should I choose?
What about sync between the different audio tracks?
Are they going to play together or start losing timing coherence?
I might look stupid asking these questions, but I think there are other newbies out there that this might help, too.

OK .. I was wondering about that. If I just connect the MAIN_L and MAIN_R that will send the same signal out to front, back and headphones? Or do I take the signal from the main out on MX6 and just attach them to all the k1lt outs? Which way would all the outs get the same signal as the Mains?
On the SB card I have the Front and Rear outputs to two different audio power amps. I want them to be the same. NOT surround sound. Just stereo. I know that the rear speaker feed is cleaner, so that goes out to my MAIN speaker amp, but the front feed I use as a vocal and computer MONITOR at my synth keyboard.

If I understand what you said, I can connect the Main_L and Main_R on MX6 to Out1L and Out1R + Out2L and Out2R + Out3L and Out3R (using the card's headphones out jack for a possible third amp) on the k1lt and that will feed the front, back and headphone speakers the same signal, in stereo, on each.

OK ... I'll go try the mapping and see what I come up with.
(grin)
I'm not really dumb ... just take a little while to catch on (wince)
Thankx for your patience and great help!!

Dennis

Russ

Basically yes, if you are not recording a track, then it does not need an input at all, right? so...

Thats pretty much it, but do not forget to click on the Record button on the Transport panel in Sonar.

MX8 would not be needed. Remember that MX6/MX8 are mixers. If you are not mixing the signals together (i.e. you are recording each midi channel to a separate track), then why would a mixer be needed?

BTW: BY default all MIDI channels use FxBus 2/3, so you would need a more advanced config to record each MIDI channel to a separate track like this (you would need to re-assign the MIDI channels to use a different FxBus in kX Router).

No, you do not have to do this (but you could if you really wanted to).

I really am not a big MIDI person, so it would probably be better for someone else to answer that.

I am not sure what you mean. I assume you are talking about MX6? The signal will be sent to whichever speaker connections that you have MAIN_L/MAIN_R connected to.

BTW: Headphones (kxlt) is for the headphone connector on a LiveDrive, so you would not use that unless you have a LiveDrive (the non-LiveDrive headphone out is usually Analog Front out)

Yes, YOU route the signals in the DSP however YOU want. If you want the same signal sent to all the outputs, then connect that source to all of the outputs.

Basically yes. Of course kxlt has to be configured correctly such that out1/out2/out3, etc are the outputs that you want to use (and again, unless you have a LiveDrive, you would not use the headphone input of kxlt (you would choose whichever card output the headphones are connected to)).

Ok, did I get everything?

At some point you really need to try these things out and thus answer your own questions (many of the basic questions can be answered by justing trying it).

BTW:
In general, you should ask seperate questions in a seperate thread. Less people are likely to reply if (for example) you ask 10 quesions, and they can only answer one of them, etc.