Multi channel record in STP v3

has any done any multi channel recording in STP3. what were your results.

If you don't want to automate the Auxes use Kontakt's automation parameters instead. Click on the Autotab in Kontakt and drag a parameter e.g. #001 to any knob in Kontakt (like e.g. the volume Master fader of an instrument). Then automate that paramter in Logic. This way you only need the Midi track in the Arrnage and not the Aux. Most Instruments fro the Kontakt library have many automation parameters already assigne, they are displayed in the Autotab.

Similar Messages

MULTI channel recording

I have MOTU Ultralite audio interface (10 in) and want to record from SOUNTRACK having access to all the input channels of the interface? HOW TO?.
I use STP 1.1

Hi . . moon,
I heartily agree with Jim on this one. My recommendation is to only use STP whenever you cant' do the job in Logic. Logic, as a rule, is MUCH more stable and far better for recording. In addition STP can import and export numerous file formats, but one serious limitation its that it can only record directly to AIFF files. Plus, STP's documentation/User Manual is where Logic's was 10 years ago (namely - lousy). I do, however, hope the Logic programmers "steal" the beautiful signal level meters from the meter tab in STP and adapt them for Logic; they're the best I've seen - - resizeable, wide range & nice response!
Holiday Wishes,
Bob

Multi channel midi recording in express

hello,
i do not know how to set up my stuff for multi channel (track) midi recording in express.
gear: motu micro lite midi interface, powerbook g4, yamaha elctronic drums dtxpress 2, edirol midi controller.
when i get this worked out i will have 3 other keyboards hooked up midi.
when i try now, my drums are bass or my edirol is drums
basic idea:
someone playing a bass line on the edirol while someone else is playing the midi yamaha drum kit.
also i would like to know if it is possible to keep my edirol connected usb running audio instruments or if i have to connect it midi to my micro lite.
i was told that it has something to do with setting up the environment.
i am midi ******** so please, if you can help, be simple.
thanks much anyone.
powerbook g4 Mac OS X (10.4.2)
powerbook g4 Mac OS X (10.4.2)
powerbook g4 Mac OS X (10.4.2)
powerbook g4 Mac OS X (10.4.2)

I'm not 100% sure about that but I think Logic Express don't allow simultaneous recording of more than one midi track. I think this is a Pro-only feature.

Waveform recording in NI RFSA phase coherent multi channel

Hi. I'm using the NI RFSA phase coherent multi channel example VI. How does the record option works? I can't quite figure out how to record waveforms properly and how to look at these recorded waveforms. Sorry, NI is still new to me. Thank you.
Solved!
Go to Solution.

Hello Kyle. Thanks for your help! I was already able to record my waveform. Now my problem is generating the waveform in playback mode. Is it right to specify the RECORD METADATA (xml document)? 'Cause when I'm doing this an error says that XML value text is illegal or out of range for type and/or format. The screenshot of the error is attached. Thank you!
Attachments:
error playback xml.jpg ‏36 KB

How can I use all tracks from multi-channel source audio with Multicam clips?

I cannot figure this out. Working in Premiere I'm prepping for a 2 camera show, with typically 4 to 6 track dual-system audio recorded by the sound guy. I can sync, and make multicam clips any number of ways–whether sync by timecode, sync by audio, or eyeballing–but whenever I edit using the MC clip, Premiere mixes down all the audio channels into a stereo single track, and I cannot pick and choose which audio track from the original to use. With 2-5 people mic'd on lavs, I need to be able to choose which audio is either cut in or heard at any given shot.
Is there any way to have control in multicam clips over which track of audio gets edited into the timeline, or even: that all tracks of audio get cut in? It's very important for the editors to be able to see waveform (which also disappears for some reason in the mixdown) and pick and choose which audio is heard?
I'm sure this might be a simple fix, but I cannot get my head around it for workflow purposes for my assistant editors. We need to use multicam, but this audio thing is driving all of us crazy. None of us can figure out how to have a multicam clip and use it with control over what audio is cut in.

True, but what I need to be able to do is have my multicam sequence in the source monitor, and cut into my working sequence with either just the audio I need from the sound guy's multi-channel WAV, OR cut in all the sound and from then I can just enable or disable what I need or don't need. But also be able to match back to the Multicam source from my working sequence.
I don't want to have to open the synced sequence, and razor blade the things I need and have to copy and paste into my working sequence in order to have the full spectrum of my multi-channel mono lav WAV files...with their waveforms visible.
Essentially, I want my Multicam clips to load in the source monitor, and the ability to cut them into the working sequence without it mixing down the audio into a single track with no visible waveform–which is what it is doing currently. Can I choose to edit in either ALL of the tracks of the WAV file along with the camera shot I want, or control which audio gets cut in?

Recommended lowest Mac Specifications for 32-Channel Recording.

I've been looking a lot into using Logic for Multi-Track recording our bands in a Live setting.
I mix on a Yamaha M7CL and just purchased two MY16-AT Expansion cards that each give us 16 Channels of Digital Direct Out output, 32 Channels output in total. To go with that, I picked up a Presonus Firestudio Lightpipe 32-Channel Interface. For the time being, I only have Logic Express 8 so I will be using that on my Macbook Pro recording 16 Channels.
My question is, what is recommended for best performance on a desktop machine when recording 32 Channels once I upgrade to Logic Studio? I'm shooting to get a new iMac, but want to be sure that I am getting a machine that wont struggle too much when I'm editing the tracks. For example, the 20" (Current Generation) iMac, 2.66ghz and 4gb ram. I'm curious how that would handle it, especially on a bit of a budget.
Any ideas or suggestions are helpful, thanks.

I've never used my Mac to record live, but I have had more than 66 mono tracks running at once in LE7 on my Macbook from an external drive without a hitch. For 32 tracks the specs you listed should be plenty of machine to handle it, IMO.

Can PCM multi-channel wav (not DD or DTS) be sent over SPDIF

<Can PCM multi-channel wav (not DD or DTS) be sent over SPDIF? --- X-Fi Elite Pro SB055A and Audio Creation Mode ---
If I understand correctly, SPDIF format can support uncompressed MULTICHANNEL wav format (within the limitations of its bandwidth) as well as the more usual DD or DTS encoded format on SPDIF.
One of the neat things about Creative's Audio Creation Mode in the Recorder section is the ability to actually record to a multi-wav format (but the sample size seems to be fixed at 24 bit). This means it is quite simple to have several mixer inputs and record a mixed multi-channel wav file in one go. Great and this works as expected with a 5. ?24 bit/48 kHz recorded test file.
My question is when one plays this recorded multi-channel wav file back, the mixer monitor (playback) shows the various 5. (say) channels with their separate content. Can this multi-channel *uncompressed* wav data? be sent over the SPDIF output as is to preserve the PCM quality? (so no DD or DTS encoding used)? I have tried this and I?ONLY seem to be able to get a STEREO (2 channel mixed down) version received at another SPDIF-in sound card (my Audigy 2ZS PCMCIA). I tried almost all selections at both ends involving SPDIF.

@? In answer to my own question, and I think I had known this a while back, SPDIF standard only supports transmission of either 2 channel PCM (uncompressed) stereo with up to 24 bit resolution per channel and up to as much as 92 kHz sampling frequency, or compressed (DD or DTS) multichannel audio up to 5..
Since each frame of SPDF is 64 bits in length (with a single sample of L and R channels plus extra bits of status in each frame), at 92 kHz max sampling rate, this translates to about 3 Mbits/sec bit rate. However most SPDF don't support the 92 kHz rate (which for 24 bit would be the stereo rate for DVD-Audio) for various copyright protection reasons.
Here is short demonstration of SPDIF transmission of 24bit/96 kHz between two sound cards with and without "bit accurate for playback" enabled:
? http://www.jensign.com/spdif/

Multi-Channel Sound Cards

Hi,
I was wondering if flash player had support for multi-channel
sound cards. Everytime I play a .swf with sound i only get audio
out of two channels.

The audio interfaces that I use are an EMU 1820 and a Tascam FW-1884 firewire device. Both are fine for what I do (broadcast quality material). I believe the EMU gives better audio quality and is certainly more robust, but the Tascam, being Firewire, is more easily movable between machines. I also like to work with real faders and it integrates well enough with Audition to be usable. The EMU has the incredibly flexible virtual mixer, but I find it hard work keeping my brain up to speed with this.
I also have other cards either inside machines or usb, but these are all just 2 channel.
I'd be surprised if anyone was using Audition for logging, as the requirements are rather specialised. I'm way out of touch with this field now, but the requirements used to be that the recorder could be left to run continuously, and users on the network should be able to pull material off at random while the recording continued. Recording is the fairly easy bit, management is the problem.
I'd have thought any computer based system ought to be capable of high quality. There are one or two low-cost or free loggers, I believe, but I've no idea how useful they are.

Multi-voice recording problem ( M-Audio USB )

Hi all...
after using my digi-cam as a pre-amp for my two radio mics, I decided it was time I bought a dedicated pre-amp / audio interface. I bought the MobilePre-USB from M-Audio. My problem is this:
I'm attempting to have the two voices on separate tracks. I basically know how to do this, by selecting each track and clicking the little red record button for multi-track recording. Then, I change one to 'channel 1 mono' and the other to 'channel 2 mono'.
In the back of the M-Audio device I have one mic plugged into the XLR socket and the other mic into the stereo mini-jack.
What I end up with is that one mic records on one track, yet the other records on both.
I have no idea why! Could it be because they should both be plugged into the XLR points?
Any help would be great.
Lee

Haven't tried this, but I think not only have I figured it out, but I'm also a little dumb.
I need to use a mono-jack into the stereo mini-jack and then it will only record onto one track as it's only using one channel
Or.. will a mono jack into the stereo jack STILL record onto both channels?
Hmmmm.... no idea!

SB X-Fi and Multi-Track Recording

I have a SB X-Fi card pre-installed on my new Windows7 computer & I'm using Cakewalk HomeStudio7 for multi-track recording. The SB card is working OK. I can record from the Line-In correctly.
However, when I record in a new audio track, the other audio tracks are also mixed in, rather than just being audible in the mix.
I didn't have this problem with a generic sound card. I'm fairly certain there is a SB setting somewhere that will disable audio from being recorded and just use the Line-In instead. No idea where to look however. I don't believe this is related to Cakewalk.
How do you keep the SB Audio output from being included in a recording? It's not connected to the Line-In & Line-In is the default with everything else disabled.
Thanks,
Mark

As I understand, the digital IO module only works with some but not all x-fi cards.
Your reciever/amplifier has to decode the DTS or other 5. encoded stream. Check to see what format your reciever can understand, and make sure that you have checked those in the Control Panel / Sound / Advanced applet. If your amp supports a particular format only (and bit depth and sample rate) you might need to make sure the card is set for that and you might possibly even need a source in that format.
The 5. channels are in one data stream, the reciever decodes it; I think they mostly autodetect the protocols they can decode. However, if you decode it in the card, you can send out the 5. analog channels to the analog inputs on your amplifier.

Including effects in multi track recording

Hi
was wondering if anyone could help me. I would like to know how (if possible) I could set a series of effects that are applied directly on a track by track basis during a multi track recording. I use a 16 channel USB interface and often record my band using +/- 10 tracks simultaneously. I now record the tracks via mics or DI outputs from amplifiers and then apply compression etc on a track by track basis after the initial data capture. I would like to be able to (for example), set a preset effect such as compressor and capture the audio with that effect applied in real time, just as I would if the signal went through a hardware compressor.
Can anyone help?
Thanks
Tony

teti_tony wrote:
so what is the purpose of the MIXER tab in multi track view where one can apply effects on a track by track basis? I am imagining that AA3 would behave similar to a hardware console where the inputs can be managed effects wise by the effects available on the board.
Are the effects in the MIXER tab for playback only? I am confused. For example, if I set certain effects to channel X in the mixer tab, what does that do for me?
The effects are present for monitoring purposes, but because they've been passed through the entire system, they inevitably gain a degree of latency that makes them rather ineffective for direct monitoring purposes by a performer. In theory, you can apply reverb into a vocal performance and send it back to the vocalist, but if you do, you have to make sure that the direct signal doesn't go as well because the delay will be totally offputting. Invariably, if you want effects to listen to whilst recording, then external hardware does a better job, I'm afraid - and that's invariably a computer limitation.
So, if you add an effect in the mixer during recording, it will only be for monitoring purposes - the track will be recorded dry, as Bob says. And yes, that's always been industry practice - you always have a clean recording to return to if you screw up the effects, and that's invariably cheaper than a new session to re-record. You have to be careful though - when you replay your track it will still sound as though the effects are applied unless you remember to remove them from the mixer channel first. So the effect acts on the channel all the time, but never on the recording.
Yes, there is a way of cheating if you have a multi-I/O card. You feed the monitor output to a spare output pair, and re-record it on a spare input pair. This way you get a dry recording, and an effected one. It's rarely worth it, though.

Setting Default Audio Channels on a multi-channel interface

Does anyone know how to set the default output channels for QT to anything other than the first 2 channels on a multi-channel interface (in my case a MOTU 828mkII)? I use the digital outputs on my interface, and this works fine wih every other app apart from QT (I select the motu in Audiomidi Setup, and then use the device driver to set the appropriate default output channels). I can do it in QT on a movie by movie basis using the Movie Properties pane, but this is irritating, and I want to set digital outputs as the default so QT sound files in a browser play back through the correct channels of the interface.
Thanks if you can help,
Rob
Dual 2.5 G5

The mic is mono. It only records on one channel. Rather than double the sound you can center it. Edit into the timeline only the track of audio with the mic. Select the audio in the timeline and press Control-period to center pan the sound.

Wanted: Multi-Channel Audio Interface for iMac & GB (USB)

I am looking for an affordable multi-channel audio interface for my iMac for use with Garageband. My goal is to playback old songs from my old computer (Atari ST) through my Yamaha gear and record the audio into GB so I can mix it. (READ: Not interested in transferring MIDI.)
My goal is to get at least 4 and ideally 6 channel/track recording in GB. I’m not sure if the multi-channel units I see, such as the Alesis MultiMix 4 or the Behringer Xenyx Q802USB send all channels to the Mac or if they mix it down to two channels first.
I’d prefer to mix all the channels in GB rather than pre-mix it. Also, I’d prefer to have 1/4” jacks; XLR would be a nice bonus. USB; since Firewire is going away and Thunderbolt is too new.
Any suggestions? I’m trying to keep this under $100. (Also, I know I can use multiple low-budget devices and aggregate them in Audio/Midi, but I do NOT want to go that route.)
Thanks.

As you imply, keeping it below $100 makes it tough.
Generally only the two channel interfaces are close to $100.
Check out Hang Times FAQ:
http://www.bulletsandbones.com/GB/Interfaces.html

Multi-track recording question

Multi-track recording question
Hey, just need some help on multi tracking using audition, hope this is the right forum?
Wanted to know if anyone has a clue about assigning inputs from an audio interface to separate tracks on a recording program (ie: Adobe audition)
I record live with my band alot, and have every amp and the kit mic'd up, but how do i make it so that each microphone is recording to a different track on the PC?
My interface is an M audio fast track ultra, if that is of any use.
Any advice appreciated! Thanks!

FreeCT wrote:
Wanted to know if anyone has a clue about assigning inputs from an audio interface to separate tracks on a recording program (ie: Adobe audition). I record live with my band alot, and have every amp and the kit mic'd up, but how do i make it so that each microphone is recording to a different track on the PC?
In Audition, assuming that you can see all of the inputs available in the Hardware Setup, you find the right-facing arrow (input assign) on each track in MV's Main view, and you click on the box it's pointing to, go to Mono, and select the input you want for that particular track. The mono ones say [01M], [02M], etc. If you want to record a stereo track, then select the Stereo option where the inputs are paired and say [01S] et seq. This would have input 1 as left, input 2 as right.
If you can't see all of the inputs available, try selecting the ASIO driver in Hardware Setup...

Important question: I have got a PXI-4472 and I am able to do single-channel acquisition. How does multi-channel acquisition work?

Look at the VI: it is quite eleborate, but the crucial point is Data Acquisition and Trigger&Gate (and, consequentely, Write to File operation).
I can add plot on the Waveform Graph DATA, this is rather easy, simply adding the channel numbers in the channel control (es. writing 0,3,7 will collect data from the three channels).
First question: is this operation of adding plots correct?
Second question: does the sampling rate dwindle when I consider multi-channel acquisition? I mean: 1 channel-->100KHz ; 2 channels-->50 KHz
More over, and MORE IMPORTANT: is it possible to set different trigger conditions for different channel
s? How can I control this operation?
Third: can you have a look at the VI as all (the appearence, the functionality, the logical sequence...). I am looking forward to get your advice. In particular:
Look at the frontal panel: what do you suggest to make it "smaller"? More tab controls?
Look at the Block Diagram: do I have to connect ERROR IN and ERROR OUT to every suVI or function that makes this connection possible (as Trigger&Gate and Write File)?
Attachments:
Start&StopTrig_SpectralMeas_style+image.vi ‏700 KB

First: yes, the operation is correct.
Second: yes again. When you consider a multi-channel acquisition, your sampling rate must be shared among the channels.
Third: when you specify a list of channels you want to acquire from, the channel considered for analog triggering is the first you put in the list.
Connect the error clusters whenever you can,in particular when you are dealing with I/O operations.
Bye!

Multi channel record in STP v3

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