Using spdif input into logic to record..

hello, I am trying to record a track in logic 9 using spdif input..
It does not seem there is an option to choose a digital input, just input 1 and 2.
I am actually recording from a second mac running logic so i have two m-audio interfaces for each computer
so i am recording a stereo mix in logic on one machine goint to logic on my second machine.
There is a reason I am doing this!
help please!

But not the standard Fast Track, that doesn't have s/pdif.
The Master Computer in this case is the playback computer, you don't need to do anything as far as setting the clock goes. The Recording computer must be set to external sync in THE FAST TRACK software, don't mess with Logic or it will get confusing, this is all done in the Fast Track software applet. The important part is that the recording computer's audio interface is synced to the digital clock coming in on the s/pdif cable (along with the audio of course.
Don't try and use MIDI to sync the two, Logic only send MIDI clock it doesn't receive it, might be able to make it work using MIDI Machine Control but the start commands will come from the recording computer not the playback one. One thing at a time, the important thing is to get s/pdif working correctly.
What are you trying to do?

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Clipping problems using SPDIF input on SB Live

.Clipping problems using SPDIF input on SB Li've?Hello everyone
I'm new to this community and I'm banking on someone being able to help me!
I'm connecting a DAT player to the SPDIF input on my Soundblaster Li've card and recording using Audacity.
Within Creative Audio HQ I've got the SPDIF volume level set to 00% (for starters, perhaps someone can explain why there is a volume control when I thought this signal was staying in the digital realm?!) and when I record in Audacity the resulting material clips with high transient sound... a piano, for example, distorts on the louder notes.
I need to take the level down to about 70% (or the equivalent in Audacity would be 0.7 - the 2 'volume' controls are connected) to avoid any clipping.
I've had the sound card years and used to use SoundForge without any problems. Any ideas why this is happening in Audacity?
I'm using XP and?I have downloaded the latest drivers and turned off the MOBO onboard sound in my BIOS.
I would be grateful for any assistance anyone can give.
Best Wishes
Steve

paush wrote:
Hi, I am trying to sample my old vinyl record collection using my Sound Blaster Li've! PCI (think that was the model - it was the original (not Value) version of the card with the seperate header for the coaxial digital and MIDI I/O). Unfortunately, I get severe clipping regardless of the recording volume on the Volume Control panel. I know that my photo pre-amp is capable of high voltage outputs and therefore wondered if anyone was aware of the sensitivity of the analogue line input in volts of this particular model? Many thanks.
Regards, Paul.
If your pre-amp is a phono pre-amp (with RIAA in-build), I suppose it's not the culprit for clipping.
How have you set the recording and source (input) levels on Volume Control and Recording Control?
Try adjust the recording level by the 'hights' (loudest outputs) of the record (you know those recordings so you can easily find those places (or you can look on vinyls surface where the darkest areas are --> loudest outputs)).
jutapaMessage Edited by jutapa on 0-23-2006 :38 PM

I have a Steinberg MI4 and a Roland Fantom X7, and am trying to use my Fantom in Logic to record audio. Need help setting it all up using MIDI cables. Please help !

I have a Steinberg MI4 and a Roland Fantom X7, and am trying to use my Fantom in Logic to record audio. Need help setting it all up using MIDI cables. Please help !

Encryption wouldn't matter except for Wifi.
While 10.2 might help, there's not much you can do on the Internet these days with less than 10.4.11
Tiger Requirements...
To use Mac OS X 10.4 Tiger, your Macintosh needs:
    * A PowerPC G3, G4, or G5 processor
    * Built-in FireWire
    * At least 256 MB of RAM (I recommend 1GB minimum)
    * DVD drive (DVD-ROM), Combo (CD-RW/DVD-ROM) or SuperDrive (DVD-R) for installation
    * At least 3 GB of free disk space; 4 GB if you install the XCode 2 Developer Tools (I recommend 20GB minimum)
http://support.apple.com/kb/HT1514
http://www.ebay.com/sch/i.html?_nkw=mac+os+x+tiger+retail+10.4
See Tom's, (Texas Mac Man), great info on where/how to find/get Tiger...
https://discussions.apple.com/message/15305521#15305521
Or Ali Brown's great info on where/how to find/get Tiger...
http://discussions.apple.com/thread.jspa?messageID=10381710#10381710
As far as Memory, that's sort of easy, find your eMac here...
http://eshop.macsales.com/MyOWC/Models.cfm?Family=emac&sType=Memory
As far as Hard Drive, it's not easy to replace the Internal drive, I'd maybe suggest an external Firewire drive to boot from...
http://eshop.macsales.com/item/Other%20World%20Computing/MAU4S7500G16/

MOTU 828 MKII input into Logic Pro

I posted about this earlier, but I think I can simplify/clarify:
I'm using 8 channels of ADAT pre-amp input connected to my MOTU 828 MKII.
Cuemix (MOTU's monitoring software) shows that all inputs are functioning properly.
But when I try to choose an input channel in Logic, I have a problem:
All 8 channels come through inputs 13 & 14 in Logic. Logic will allow me to select inputs 13 through 20 (they're not grayed out), but there's no audio coming to inputs 15 through 20. And, yes, I'm pressing "record" to enable the track.
I should be able to assign inputs as follows:
ADAT 1 = Input 13
ADAT 2 = Input 14
ADAT 3 = Input 15
ADAT 4 = Input 16
ADAT 5 = Input 17
ADAT 6 = Input 18
ADAT 7 = Input 19
ADAT 8 = Input 20
Any ideas on what's happening here?

The reason it is 2 out is because Logic doesn't recognise the names of the inputs. They are simply numbered. So, the front 2 inputs on the MOTU are inputs 1 and 2 in Logic. So the next input in Logic is 3, but the next input on the MOTU is on the back..... which (rear of MOTU) is input 1..... and on it goes.
Logic 4, Motu (rear)2.
This got me at first until I realised that Logics input chanels were just numbered straight through, not differentiating between front, rear, analog, ADAT etc.... I think inputs 11 to 18 in Logic correspond to ADAT 1-8 on MOTU.

No audio input into logic 9 using Apogee Rosetta

Hi there, I've just upgraded to Logic Pro 9 and realized that all the inputs (1-32) are greyed out and unselectable when I go to make an input selection.
I'm using an Apogee Rosetta 800 and didn't have this problem on Logic 8.
Any ideas how to solve this problem?
Thanks in advance.

ADPhoto wrote:
Yes it is checked on. Sorry that wasn't clear in the last post. I tried checked off as well. But from what I have read it should be on.
Software monitoring only needs to be on if you wish to monitor through Logic's audio engine. Most audio interfaces also have a direct Monitoring function which allows you to monitor the inputs of the audio device. Direct monitoring virtually eliminates latency (except for converter latency which is minimal).
So... you've created a -MONO Audio- track in Logic? You then select the track and can see the I/O channel strip on the left hand side of the page.
There are two I/O buttons right above the meter, the bottom is output and usually set to 1/2. The one right above sets the input channel of the Audio device... make sure Record is enabled then select that button, from the drop down choose Input, how many do you show as possible inputs 1, 2, 3, 4?
Play the bass and select each input until you show some input level.
pancenter-

No Audio input into Logic 9 using Firewire Solo interface

I am trying to get audio (bass guitar) into an audio track in logic. I am getting a signal to the Firewire Solo panel in preferences. I have Link and out 1/2 selected in the audio interface panel. In Logic audio preferences I have selected Firewire Solo as the input device. I create a new Audio Track and I get nothing. I have tried every option when creating a new track. I have read that many people have had the same problem in other forums but I have yet to hear a solution. Any help would be appreciated.

ADPhoto wrote:
Yes it is checked on. Sorry that wasn't clear in the last post. I tried checked off as well. But from what I have read it should be on.
Software monitoring only needs to be on if you wish to monitor through Logic's audio engine. Most audio interfaces also have a direct Monitoring function which allows you to monitor the inputs of the audio device. Direct monitoring virtually eliminates latency (except for converter latency which is minimal).
So... you've created a -MONO Audio- track in Logic? You then select the track and can see the I/O channel strip on the left hand side of the page.
There are two I/O buttons right above the meter, the bottom is output and usually set to 1/2. The one right above sets the input channel of the Audio device... make sure Record is enabled then select that button, from the drop down choose Input, how many do you show as possible inputs 1, 2, 3, 4?
Play the bass and select each input until you show some input level.
pancenter-

SPDIF straight into macbook pro/logic?

I've been trying to run an effects unit with spdif out into logic on and off for some time now, with no success. I had been using a firepod, which theoretically has spdif ins, but i was never able to get audio into logic with that method.
Is there a way to get spdif data directly into my macbook pro and bypass the audio interface entirely? (I'm a little new to digital audio recording stuff.) I'm going to be returning the firepod to it's owner soon anyway and picking up an apogee duet...which as I understand does not have SPDIF inputs.
The piece of gear I'm trying to interface with is the fractal audio ax-fx. I believe it needs to be the master on the spdif clock and only outputs at 48kHz.
Any ideas?
Thanks!

chorleyman wrote:
Oh, OK, I stand corrected Pancenter - I had no trouble doing what I did with it and I have s/pdif connections on domestic equipment (TV/hifi) that require nothing more than plugging them together! Presumably the toggle in the Firebox/Firepod control panel provides the necessary clock sync but, as I stated, I don't know how the digital in/out on the MBP works (I was hoping someone like yourself would clarify!).
CM,
You may have been in luck and the digital clocks were so close together that you didn't suffer any noticeable artifacts, either that or your units auto-synced. If the digital clocks on both units are very close to the same frequency digital noise can be minimal, still, the audio has lost some integrity.
Let's say this is taking place at 48kHz, unfortunately each unit's digital clock is not exactly 48kHz. One of the most noticeable artifacts is a slight ticking noise, another will be a small pop, tick, or sizzle that occurs randomly.
The Master device is almost always the device sending the digital signal, the receiving device is set to external clock and runs off the clock signal embedded in the s/pdif, ADAT or Optical stream. The other way to do it is using a Word Clock generator so that all studio devices sync to the same clock. That's usually the best option as it's the most stable, jitter free method of sync.
pancenter-

Recording Streaming Audio Directly into Logic 8?

Is it possible to record streaming audio directly into Logic 8. I would like to grab some sound bites from web sites and various other places and lay them down as Audio tracks in Logic 8.
I know WireTap Pro will do it for $30 but I wonder if I could just do it in Logic.
Again, thank you for your time.

Bee Jay wrote:
Or you could also use the loopback trick and just plug your interface's outputs >into it's inputs - be careful of your routing to avoid feedback though...
Hi Bee Jay .. I wonder if you would mind advising on this setup:
Normally I hear everything that comes out of the computer via the following route:
Out of computer via Firewire and into Motu828
Then out of Motu 828 and into Mixer channels 1 & 2 which are set to Monitor only. ie I hear all audio ( including streaming audio from computer) through my monitor speakers and the signal does not go back into Logic.
All the other mixer channels (3 - 16) I use for recording. Mics and line inputs routing these back into the MOTU (and then the computer) by buttons on the individual channel of the mixer. ie If the button is up on a particular mixer channel it comes out the speakers and does not record in Logic. If the button is down then that channel records into Logic (via a loom into the MOTU).
I was warned on no account to press the buttons down on channels 1 & 2 ( because those were the main outs to monitor Logic's signal ) ...
But that if I *switched off the channel output in Logic channel strip* - then I could press down those buttons on Channel 1 and 2 and route whatever audio was coming out of the computer ( iTunes, You Tube etc) back into Logic to record.
I did this some time ago and it worked ... but for some reason, now even though I have the Logic Channel outputs switched off, I still get the most horrendous howling feedback if I try to record Streaming Audio or iTunes.
If you or anyone else has advice on what would be causing this, I would be most grateful
ms

PreSonus Firebox SPDIF input issues

Hello all,
I'm hoping you folks could help me on an issue I've been having with getting the input chain correct to record a SPDIF input signal. I've searched this forum with no other postings found. I've also emailed PreSonus tech support without much luck. The emails I sent are posted below. I'll refrain from posting the replies as they are the property of PreSonus. I'm not looking to bash them. I just want to get on with recording. Please read through them if you feel you could be of help to me.
Thank you,
1st email...
Hello,
I'm an running the following equipment;
PreSonus Firebox
Logic Pro 7.1
Apple 12 inch powerbook running OSX Tiger
Focusrite Voice Master Pro
I recently purchased the digital A/D card with SPDIF for the Focusrite Voice Master Pro (preamp / channel strip). For the life of me, I have not been able to figure out the recording chain for the digital SPDIF input coming from the Focusrite Voice Master Pro. The Firebox manual shows a selection is needed for Windows XP within the Control application. However, non exists for OSX. I'm not a new user to DAW systems. I recently upgraded from a MOTU setup and used SPDIF to / from a DAT recorder without issue. Considering how simple it was to use digital recording with that set up, I know I must be just missing something here. But, after a week of reading the manuals and reading the Logic online forums, I'm still stumped. Please advise and help.
Thank you,
In the reply, I was told I had to download the control app for OSX and select inputs 5-6 within the logic channel strip. I did so and still no go.
2nd email...
Thanks for the reply. I did download the app and tried to use 5-6 input without luck. No signal, bummer. I guess I have more than one issue? I'll have to do some testing by changing cables and running the signal from my DAT player instead of the Focusrite. Maybe I have a bad A/D card on the Focusrite. I'll try running the Focusrite into the DAT as well. Is there any way to test and/or monitor the SPDIF on the Firebox (without Logic) for sync lock and signal? I tried installing the Cubase software included as a test, but I get an error message. So, no go there.
In this reply, I was told I should download the Firebox Mixer and a patch for Cubase. The mixer came with the unit so I really don't see the need to download it unless there's an update. I really don't have a need for Cubase. I just wanted to install it in order to use it as a test.
3rd email...
Hi,
Good news... I was able to send output and record to my DAT player via SPDIF. And, the Focusrite also recorded to my DAT player via SPDIF.
Bad news... No input via SPDIF via the Firebox.
I see that there are many options to route the output signal though the mixer, but no "real" input chain settings. There's a lack meter or peak monitoring on the mixer or the unit. There's no way to be certain the correct sample rate is set and/or received so that I can be sure a digital lock is established. I set Logic to input 5-6. Nothing. I can only assume that when a SPDIF signal is received, it's automatically set to 5-6. Is this correct? If so, something must be wrong with the unit. Perhaps you can tell me (or send me a screen shot of) the correct mixer setting I should be using so that I may test the Firebox.
To be honest, I'm about to trash this unit. I've spent several hours trying to figure what's wrong here. I find the documentation woefully inadequate. It's integration lacking. The fact I had to download the undocumented control software to get the SPDIF capability that was to already to be included with the unit as advertised was not what I was expecting. Partially against other manufactures products I've used. I do appreciate your help in this matter. At the same time, if I have to spend much more effort on this, I'm more likely find myself buying a better box. It will not be a PreSonus.
I sent that email 2 days ago and never got a reply.
I really don't want to buy something else when I'm likely to be just missing something. Please help.
Thanks,
Tim
G4 12inch Powerbook Mac OS X (10.4.4)
G4 12 inch Powerbook Mac OS X (10.4.4)

Well.... I did download that control app and I finally got the SPDIF input working. That's a different version than the 1st control app I was told to download just days before. WHATEVER. I'm a happy guy now. Wheee!!! The funny is/was that's the same download the presonus tech told me was the the mixer app. That's why I didn't download it right away. The 1st couple of times I used it caused some audio shuttering, but then that has gone away. Hopefully, that's not a new issue.
Good luck to all.
Tim

How to synchronize video camera with logic for recording?

Hi guys,
I have a Sony VX 2000 and logic pro.
I'd like to do a multitrack audio recording of a musical theatre performance and simultaneously record video with two cameras.
I want to record the sound into logic pro, that way I can mix the sound later (multitrack audio).
At the same time I want to record video with the cameras.
There may be 100 takes or more....
What is the best way to synchronize logic with the video recording?
I don't want to synchronize each audio take with each video take later, there must be an easiser way!
Is it possible to start recording of video and audio at exact the same time?
Do I need any additional hardware?
Any feedback highly appreciated!
Thanks Eddie

Have the drummer make a 2 pop (click) at the beginning of every take.
Set the camera to 16 bit audio and you can import the camera audio into Logic.
Record multitrack into Logic at the same time.
You can align the takes to the audible click.
Sorry, It's the only thing I can think of,
without having very expensive externally synced cameras.
Logic doesn't record video, it plays it back.
So making some type of slate is the way to go.
It will be a pain to manually align all that stuff.

I am running Logic Pro 9.1.7, and I use a Roland RD700GX keyboard to input midi data into Logic. How do I set up Logic to record the Roland's own internal Grand Piano sounds into Logic, while retaining the ability to make midi edits? By Eggie

I am running Logic Pro 9.1.7 on a mid-2011 Intel iMac running OS10.7.4. I use a Roland RD700GX as my midi input keyboard. How can I set up Logic to record The RD700GX's own internal Grand Piano sounds, while retaining the ability to take advantage of Logic 9's midi editing capabilities. I am a relative newcomer to Logic, and I haven't been able to find an answer in The Logic 9 User Manual. A step by step procedure would be much appreciated. Thank you.
Eggie0379

Create a new External Midi Track, set the MIDI port to the one being connected to teh RD700. Now,create a new Software Instrument track. Select the external instrument plug-in. Set the MIDI destination and the audio input inside the Plug-in. Now, you can record the MIDI notes and hear the sound of the RD700. Once you've editied, you can bounce the track.
Best,
DaCaptain

What connectors do I need for my ipad so I can connect it to my apogee duet 2 interface essentially using the ipad as an instrument utilizing a few of the synth and drum machine apps to record tracks into Logic on my Mac?

Ok, so I completely feel like this is a stupid question for those of you already in the know and recording tracks onto your Computers DAW from instrument apps on the iPad and other i devices. I can not seem to find out exactly what connector or unit I need to be able to connect and use my ipad as a instrument to record tracks onto logic. I have an ipad2 that I want to connect to my apogee duet 2 interface which is obviously connected to my Mac. I know Logic comes with plenty of drum and synth options but I still want to be able to use beatmaker, jam up pro and the sunrizer synth apps on some of my tracks. I do have the camera conversion adaptors. So my question is, do I just run from the camera adapter in the ipad into the midi port on my apogee interface? (I'm new to and still learning about midi) Or is there a special 30 pin cable I need to get that runs from the ipad into the XLR or quarter inch in the break out cables on the apogee? I did find one cord out there that plugged into the ipad and had left and right XLR connectors but my duets break out cable only has two inputs each is their own channel. Also , if the cable I need isn't midi and does connect to the breakout cables, do I set it as an instrument or as a mic in the apogee software? Or am I completely way off base and needing something different all together. Any help would be much appreciated!
I have a fare amount of experience when it comes to recording, set up and the flow of standard instruments, using external hardware and plugins. But the last few weeks I spent a lot of time programming some great beats in beatmaker on my ipad and I want to track them into Logic. It seemed like no matter how I worded the questions when researching what I thought was and should be simple to figure it out, it just took me to answers I wasn't looking for. Thank you ahead of time. Like I said I'm just starting to get out of my comfort zone and learn about midi and the more digital side of things. So sorry about any ignorance concerning that side of things.

The new Apogee Duet for iPad and Mac, is compaible with iOS devices and includes a USB Midi input. This sounds like the perfect solution for your workflow.

How do you Control SPDIF input levels in Logic Pro using Apogee Ensemble

Hello,
Question. I have an Apogee Ensemble audio interface hooked up to my Macbook Pro. I have a Motif XS8 that I record audio from through SPDIF.
I am able to get sound through to my audio track once setting the inputs in Logic, however the levels are not even close to halfway. I can not figure out how to fix adjust the SPDIF level to make it hotter b/c I know once I add virtual tracks and vocals through my preamp, the audio from my motif will likely not be hearable.
Please let me know.
Also, when I record an audio track is says about 15 minutes remaining to record. Does that mean I have 15 mins of audio for each track or for all my recording period???

Not owning an Apogee Ensemble, maybe someone else who does has more definitive info can jump in, but do not confuse the lower level of the SPDIF input with "lesser quality". It's a digital connection, with no analog conversion.
-18 dBFS on a digital scale is the same as 0dB on an analog scale.
I think what you're experiencing, is the fact that so many people think they need to get the meters in Logic closer to 0dB, which is actually NOT what you want (when staying ITB). This thinking is left over from the days of analogue recording, or the early days of 16 bit digital recording.
Turn your other sources in Logic down, and you'll reap the benefits of not overloading the 2 buss, allowing the plug-ins to do their computations without distorting, etc... Then you can bring the over all level of your mix back up, in mastering.
If the Motif is considerably lower in volume than that, you could always insert a gainer plug-in on the audio track.
As to the 15 minutes, that is what Logic has pre-allocated for recording to your hard drive. It "re-sets" each time you go into record, so it's not saying "you only have 15 minutes of record time available". It's saying, "you have 15 minutes of hard disk space allocated each time you go into record, based on your sample rate and bit depth settings". This can be changed in the Audio pathway, but it's recommended to leave this number as low as possible/necessary, to avoid disk fragmentation.

I am using logic to record a podcast and using soundboard to use as a playback for music and sound effects. I'd like to soft patch sound board into logic, and still use my interface for recording audio from the other members of the podcast.

I am using logic to record a podcast. I have a presonus firestudio mobile for my interface so I can record several people at the same time. I'm also using Soundboard for a playback system for music and sound effects. I'd like to know how to soft patch Soundboard into Logic while still using the Firestudio interface to record the other members of the podcast. Does anyone know how to do this? and if so can you tell me how to do it? Thanks.

Thanks. Now, since Apple failed to provide an audio in jack for their new and improved iMacs (mine has one), all I can suggest is that you contact AppleCare, since you have 90-dqy free phone support. If need be, you might reconsider your purchase, but that's for you to decide.

Latency, Recording incoming Audio into Logic,... I don't understand

I'm trying to clear up my understanding (or misunderstanding) of latency of incoming audio into Logic.
As a test, here's what I did.
Set-Up
1. In Preferences/Audio/General, I set Plug-In delay Compensation to off.
2. In Preferences/Audio/Core I set the I/O Buffer Size to 32 (is this necessary?, to go that low?)
3. In Preferences/Audio/General I turn on the Low Latency Mode On, set Limit to 0ms (Zero Milliseconds)
4. Created a 1/4 note click track with Klopfgeist.
5. Just for reference, I converted to audio. Opened in the sample editor and it is indeed, exactly on the beat.
6. Set up my Roland Fantom as a Midi instrument. Channel 10 to trigger a Side Stick sound.
7. Create another 1/4 note click track.
8. Recorded the Rolands Output into my MOTU 2408 interface into Logic.
9 Opened up the waveform up in the sample editor, zoom in and notice it is obviously late.
10. Move the Anchor point 6 ticks to line it up,
Therefore, the incoming audio is off by 6 ticks.
Is it not possible to get 0 latency? I read the chapter in the manual but I believe this wasn't made clear, i could be wrong.
I have a guitarist coming over to record on my record in a week, I don't know how to deal with this except to move his entire track
by 6 ticks. ?

That is not really the type of latency that's affected by your audio I/O buffer setting. A DAW system has different types of latency, MIDI latency, latency caused by some plugins, AD/DA converter latency..etc. The six tics will probably remain the same or close to it if you're buffer is set to 64, 126, 256...etc.
Why do you have "Low Latency" mode selected, do you know what it's function is?
The I/O buffer setting affects two main aspects.
The response of the system when you play a virtual instrument, and...
...real-time monitoring of an audio input track. (this is an audio signal that passes through Logics audio engine and can use live effects)
The example using the Fantom requires MIDI sent to the keyboard and the audio into the MOTU passes through the AD converter.
What you want to set is the Record Delay so that your audio records in time, however... it needs to be set using a loopback test.
Use a 1/4 note recorded audio click (side stick) that you've adjusted to be perfectly on the beat copy it a few times. Use a cable from the MOTU's output to an open channel's input, record the click, compare.. now adjust the "Recording Delay" until the recorded click matches the first click.
Recording delay is on the Preferences/Audio tab.
pancenter-

Using spdif input into logic to record..

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