Help! Problem with recording on time

When I record something with logic the program always places the file slightly ahead of where it was recorded. Every clip i record i have to zoom in and correct the timing. Does anyone know what could be causing this?

hello ghostdays,

 

Under Preferences > Audio > Devices, is your Recording Delay set to "0 samples"?

 

Also, what audio interface are you using? Please add your Logic version and system info to your signature: Read Me Before Posting - Forum Guidelines (#5)

Create an audio file with, say, a sidestick on quarternotes.

Play it through an output, patch that output to a new input and record that to a new track. Do the clicks line up ?

 

If not, tweak Preferences>Audio>Devices>Core Audio>Recording Delay until they do.

 

Christian

Recording Delay

 

ski once wrote a detailed procedure for how to adjust this in a precise way: "HOW TO DETERMINE AND SET THE RECORDING DELAY."

 

I've never tried it, but I assume it works.

The "ping" function of Logic's I/O Plug gives me the same result as the various record and count samples or null tests.

That's good to know, thanks for mentioning it. I like doing things the easy way!

Hi, I noticed this problem when I tried to record some guide vocals on a project.

I tried to follow ski's procedure without much luck. Can't find the "PDC" box in audio preference's, Don't know what Logic I/O plug-in is. When I loop input to output, record meter doesn't work etc.

In Preference's recording delay was set to -421. Did logic 9 calculate this and set these parameter's for me?

Also what does the "I" but do on audio tracks? I assume it monitors at the channel Input, but I still hear whatever effects I have on the channel when it's pressed.

I tried to follow ski's procedure without much luck.

 

I think redlogic has pointed out that it might be easier to use the Ping button in the I/O Utility.

 

Can't find the "PDC" box in audio preference's

 

When ski uses that term he means "plugin delay compensation." Which is another term for plugin latency compensation. These topics are discussed in the user manual. The chapter entitled "Working with Plug-in Latencies" starts on p. 1197. You can find that pdf at documentation.apple.com.

 

Don't know what Logic I/O plug-in is.

 

At the same location, you can find a pdf called "Logic Studio: Effects." The I/O Utility is discussed on pp. 263-264.

 

what does the "I" but do on audio tracks? I assume it monitors at the channel Input, but I still hear whatever effects I have on the channel when it's pressed.

 

That's not a bug, it's a feature. This feature is explained in the user manual on p. 435:

 

It is important to note that the effect plug-ins are monitored but not recorded, which can be useful during a recording session. For example, during vocal recording many singers prefer to hear their performance with a little reverb, but the track is recorded dry—without effects.

Go to Logic Pro > Preferences > Audio > General. The Plug-in Latency section is in the middle of the page. It's good to know where that is, but I don't think that's your problem.

 

I/O Plugin: Click on an insert slot and then Utility > I/O.

 

The "Ping" version of determining recording latency

(This describes using input 4 and output 4 of your audio interface, but you can use any unused in/out.)

 

1. Open an Empty Project from the Factory Templates.

 

2. Create an audio track.

 

3. Change the Output of the Audio Channel Strip to Output 3-4. This automatically creates an Output 3-4 Channel Strip.

 

4. Change the Output of the Audio Channel Strip back to Stereo Out. This is VERY IMPORTANT

 

5. Click on an insert slot of the Audio Channel Strip and open the I/O plugin.

 

6. Set the I/O plugin's Output and Input to 4.

 

7. On your audio interface, use a cable to connect Output 4 to Input 4. Don't engage any fancy digital loopback stuff. Use an actual cable that you can touch and feel with your hands.

 

8. On your audio interface, make sure Input 4 and Output 4 are not muted and their faders are up.

I'm not familiar with the Phonic 808, but in the software Control Panel for my FireFace 400, there are faders for its physical Inputs, physical Outputs and DAW (Logic's) Outputs. On the Fireface, Logic's Output 4 must be routed to the Physical Output 4 of the unit.

 

9. In the Mixer Window click the "All" button.

 

10. In the I/O plugin, click the "Ping" button. If everything is set up correctly, it will register the ping on Audio 1, Stereo Out and Out 3-4. If there is latency (positive OR negative), it will be shown in the Latency Offset window of the I/O plugin. For my system, it shows +2 samples. So, I go to Logic Pro > Preferences > Audio > Devices > Core Audio and enter -2 (minus 2) in the Recording Delay window.

[Edit: My new setting is "0" thanks to improved RME drivers. ]

Edited November 20, 2010 by redlogic

Thanks, redlogic. That was perfectly clear and easy to do.

Thanks Redlogic. I'm finally getting back here after chasing Ski's ten page thread for a few days.

 

After finally understanding what we're needing to accomplish, I routed output 1-2 to input 3-4 on the 808 (tried to go 1-1,2-2 but the interface complained even after software monitoring was turned off. Anyway, I played an apple loop on the first track, recorded it to the second track and kept changing the delay numbers in the audio preference panel until the tracks lined up. I think that pretty much accomplishes the same thing eh? Anyway, I ended up with a number of -11 instead of the -421 that was originally there.

I'll try your ping procedure to verify now that I know more about the test.

 

Haven't tried to record anything yet, but when a friend listened to my first project, he said it felt goose-egged which is his definition of loose timing. And I never could get it to technically feel right let alone getting to mix-down.

Just ran RedLogic's procedure and ended up with +11 which matches quite nicely with my results from previous post. Thanks all. Now I can get back to making music.

Just ran redlogic's loopback test using the I/O plugin... this is MUCH easier, faster and more accurate than the old loopback test (that Ski developed for Logic 7). So if you have LP 9.0 or newer, this is the way to go to find your recording delay. If you have an earlier version of LP, then use Ski's method - it's your only choice. (Older versions of the I/O plugin don't have the ping and latency calculation found in LP 9.0 and newer.) Edited September 21, 2010 by ERO

Just tried to RedLogic's I/O loop test and I got a result of +58 samples. So I entered -58 as my recording delay setting.

However I tried to confirm this setting by having a click file on one audio track, routed to the in's on another audio track to see if the results would line up.

This is on max magnification. Where should I be taking the start of the waveform from, and where is the slight registering in the beginning of the waveform irrelevant from?

Audio file

Not sure how you actually did your test, but the best way to check if your recording delay is right would be:

 

1. Open a new empty LP project and set up a couple of new audio tracks; put a short drum loop on one of the new tracks.

2. Connect a cable from your main output to one of your inputs; set the input of the second audio track the physical input you plugged into.

3. Record-arm the second audio track, and hit record. The first audio track will play back, go thru your outputs into your inputs and be recorded on the second track. Record a few bars.

4. Now, zoom in and check the transient of the first hit of the drum loop. Compare both tracks. If your recording delay is set correctly, they will line up perfectly. If they don't, then re-do Redlogic's PING procedure to find the correct recording delay.

 

Hope that helps...

Not sure how you actually did your test, but the best way to check if your recording delay is right would be:

 

1. Open a new empty LP project and set up a couple of new audio tracks; put a short drum loop on one of the new tracks.

2. Connect a cable from your main output to one of your inputs; set the input of the second audio track the physical input you plugged into.

3. Record-arm the second audio track, and hit record. The first audio track will play back, go thru your outputs into your inputs and be recorded on the second track. Record a few bars.

4. Now, zoom in and check the transient of the first hit of the drum loop. Compare both tracks. If your recording delay is set correctly, they will line up perfectly. If they don't, then re-do Redlogic's PING procedure to find the correct recording delay.

 

Hope that helps...

 

I did exactly that although instead of using a drum loop, i used a click sample of the very first beat of bar two. Using RedLogics ping test I had a result of 58 samples, but when i entered -58 as my recording delay, I got the above result shown in the screen shot.

If 58 was the correct value, the above waveform would be perfectly aligned to start at bar 2, yet as you can see there is waveform BEFORE bar two.

It looks to me like the two tracks are perfectly lined up... your source doesn't hit right on beat 2, so why would you expect your recorded track to hit right on beat 2? Try the test with a source that hits right on beat 2, and I'll bet your recorded track will hit right on beat 2.

 

If you do the same test with recording delay set to 0, you'll see that the recorded track is late relative to the source. All this process does is try to compensate for the recording delay. So if the two tracks match up (in time) using a -58 sample delay, then you have set your recording delay correctly.

It looks to me like the two tracks are perfectly lined up... your source doesn't hit right on beat 2, so why would you expect your recorded track to hit right on beat 2? Try the test with a source that hits right on beat 2, and I'll bet your recorded track will hit right on beat 2.

 

Sorry the screen shot shows the result of a stereo track recorded. Not my source audio track at all.

I ran Reds test, got 58 samples, and i entered -58 as my recording delay.

I ran the audio loop test, and open in the samples editor expecting to see my new recording slap bang on Bar 2....

Its not..... its early as you can see. The screen grab is the resulting stereo audio file I recorded.

You need to compare the recorded track with the source track, not with the grid.

This is my source track. One audio click pulse. Actual waveform starts one sample after the grid as you can see.

What does it look like on the Arrange page? Zoom in but show both tracks, one above the other.

What does it look like on the Arrange page? Zoom in but show both tracks, one above the other.

 

This is what i get. Recorded with PDC off, software monitoring off, metronome off etc.

Using Red's ping test i got a result of 58 samples. So entering -58 as my recording delay value, i now get this.

I've expanded the start of the recorded region to show before the start of bar 2. Is the slight waveform that is showing before Bar 2 of importance?

This looks pretty close to me... I would try two more things:

 

1. Try -59 samples to see if it lines up better. Just put the playhead right over the transient and see if the two waveforms line up better [at -59 vs. -58].

 

2. Put a Gain plugin on the second track, reverse the phase, and then play the two tracks back together. You should get nearly complete phase cancellation.

Edited June 29, 2010 by ERO

2. Put a Gain plugin on the second track, reverse the phase, and then play the two tracks back together. You should get nearly complete phase cancellation.

 

It's like it almost vanished. I get a very thin faint pulse.

 

What about the waveform that's showing before Bar 2? Of no importance then?

What about the waveform that's showing before Bar 2? Of no importance then?

No. It's your DA - AD stages trying to follow the waveform as closely as possible.

 

Christian

It's like it almost vanished. I get a very thin faint pulse.

 

This is additional proof that you have indeed found the correct recording delay.

This may be a subject for another thread, but when I recorded output 1-2 to 3-4, I noticed the waveforms were not the same. Though it was easy enough to compare the two's timing, one wave form is somewhat different than the other. Is this "coloration" from how the interface samples, do more expensive interfaces have less coloration of the waveform, has anyone else noticed this. It seems if both source and sink were set to unity gain, the waveforms ought to look the same, no?

Here are some notes form Ski's original post, that I did not see mentioned here:

 

HOW TO DETERMINE AND SET THE RECORDING DELAY

 

STEP 0 -- very important!

 

• Turn software monitoring off

• Turn the metronome off

• Set the recording delay value to zero

• Set PDC off

• Make sure you have no plugins anywhere.

 

When you are doing this loopback test, what does it solve? JUST the delay that would be induced by, say, recording guitar to a session where you are playing along with drums or something that is already in the session?

 

Do the loopback test results change if you change the buffer size?

 

Do you still have to use PDC to compensate for plugin delay as you build out your session with software instruments and plugins?

 

Are the ping test results the same as logic's calculation of "Rountrip latency"?

Where is the latency offset window???

 

I have a firewire solo and it doesn't have input 4 so i used input 1 and output 1 but when i click ping there is no results anywhere...

 

Can someone please help me with this