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Ski's Recording Delay Test


Eric Wikman

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I've posted the steps that Ski wrote ( from the link in the other thread) for testing your I/O latency. I hope Ski doesn't mind. If so, I will edit. I didn't want to hijack his other thread either. I really wanted to know where my recording where going.

When I read the steps, I thought it would be easy.

 

I did this test today, and although I had some troubles, I think my Metric Halo delivers my audio at neg (-)1. So I guess my driver puts it early. Not sure how this all works. :)

 

A couple questions. ( I guess for Ski).

I used 128 buffer. Does this matter?

 

Also, I assume PDC off, though no plugs are used except the gain plug.

 

My problems were:

when pressing play in the sample editor only the track opened (Y) would play, so I had to stop, close the editor, hit play in Arrange window (both tracks playing now), open the Y track again, and now my zoom in settings were lost. Every time! Why is this? I am still kinda new to Logic. So I hit stop, which now played the track opened in the editor.. on top of the 2 tracks playing in the Arrange. UGH. clumsy.

 

Second issue was it seems that for my first test, the wave form beginnings / starts looked different from the original slightly. I also noticed that my new recordings were 1 db louder than the orig. So this made it so my files would not cancel out completely. Now I'm left asking why my files are changed at all during this short round trip recording. I reset my Metric Halo, double checked all levels etc. Where is this distortion to the waveform coming from? With the sine wave test, both waves looked the same though.

 

I also had trouble moving the anchor around. I realized when grabbing & moving it, it looks like it's not moving against the time-line, but sliding the file behind as you go. Just odd to me. I had to reveal some of my "Y" recording at the top of the file, then move the start point to the left. Then I had go to the arrange page, and select to place my recording back to the original position. I was then able to back into the sample editor and move the anchor 1 sample to the left, to get the best canceling This was tricky though, and I messed up a couple times.

 

So instead, I was able to look at each file (both opened in the sample editor), zoomed all the way in... and set the "marking line" to the very sample at which the waveform started. I opened both tracks and could see that my recorded tracks were 1 sample before the orig every time. (See attachment)

The files I used were a Stereo Rhodes, A sine wave, a drum groove and a full song. Sometimes I made the orig file region boundry start point right in the middle of a chord, so the wave would show an abrupt start for visual aid.

I will test again to confirm or adjust my methods if I need to.

Also, with plugins added I guess.

thanks for your help.

 

________________________________________________

 

0. Turn software monitoring off and set recording delay to zero.

 

1. Import a CD track (or any stereo track of your own, preferably something with sharp transients at the top, like drums) into track 1, audio channel 1 of the arrange window.

 

This track is going to be called "X". Align it to start at bar 2.

 

2. Set up track 2, audio channel 2 in the arrange window to record from inputs 1/2. This is the track you're going to record your looped-back audio on.

 

3. Make sure the fader levels for both channels (tracks 1 and 2) are to 0 dB and set their outputs to 1&2

 

4. Use patch cables to connect outputs 1&2 of your audio interface to inputs 1/2

 

5. Start playback at bar 1 and go into record a little before bar 2 (punch on the fly works well for this). The recording you're making will be called "Y". You only need to record about 20 seconds of material max.

 

(BTW, this little bit of air in the front is for troubleshooting purposes in case this procedure doesn't work.)

 

6. Now take track 2 out of record and insert the Logic > Helper > Gain plug, setting L & R channels to be out of phase.

 

7. Reduce the level of output 1&2 by 6 dB (to prevent clipping at the output in case your tracks are loud)

 

8. Open "Y" in the sample editor. Zoom ALLLLLLLLLLL the way in to the anchor point. Set "view" to "samples".

 

9. Click/hold on the anchor point, being careful not to move it. Note the numbers that appear in the upper left hand corner of the window. Write down the bottom number.

 

Since latencies cause re-recorded audio to be late, you're going to want to advance the anchor point to the right to find the null point, as follows....

 

10. Play back your tracks. Move Y's anchor point to the right one sample at a time until you start to hear the sound thin out. Start/stop Logic as needed. As you approach the null point you'll hear the sound get thinner and thinner.

 

At some point you will hear the sound get very thin, and then, moving one sample to the right later, cancel completely. When this happens, click and hold on the anchor and write down the number on the bottom.

 

11. Subtract the first number from the second number. Then put a "-" in front of it. THAT's your recording delay value; set it in your audio prefs.

 

12. To confirm that this is the correct number, delete "Y". Make a new recording of your original track "X" onto track 2. This is going to be called "Z".

 

13. If the Helper > Gain plug is still active (putting track 2's audio out of phase), when you play back both tracks now you should hear silence. To confirm, bypass the plug and you should hear your original track 2x as loud.

 

If it's close but still not perfectly canceling, try adjusting your recording delay +1 or -1 from the number you set it to and repeat steps 12 and 13 again.)

waveformsample2.thumb.jpg.7a4ccb292cfc34097e2a212fb40b4f5e.jpg

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I hope Ski doesn't mind.

 

Nope! 8)

 

I used 128 buffer. Does this matter? Also, I assume PDC off, though no plugs are used except the gain plug.

 

Buffer size shouldn't matter. And regarding PDC, theoretically, as long as you don't have any latency-inducing plugs in the song you're using to do this test, it shouldn't make a difference if it's on or off.

 

...my zoom in settings were lost. Every time! Why is this?

 

That's the way Logic's been since day-1. The way around this is to leave the sample editor window open but make it go to the background by clicking on another window (like the arrange page). To get back to it you can select it from the Window menu. Or you could resize the arrange window so that the sample editor window peeks out from behind and easily re-accessed by clicking on it. Finally, there's a key command called "select next window". If you had only the arrange and sample edit windows open, for example, you can use this key command to effectively toggle back and forth between those windows.

 

Second issue was it seems that for my first test, the wave form beginnings / starts looked different from the original slightly. I also noticed that my new recordings were 1 db louder than the orig. So this made it so my files would not cancel out completely. Now I'm left asking why my files are changed at all during this short round trip recording. I reset my Metric Halo, double checked all levels etc. Where is this distortion to the waveform coming from? With the sine wave test, both waves looked the same though.

 

I seem to recall that someone on the Apple/Logic forum reported this as well. I'm not familiar with their stuff so I can't be of much help there.

 

I also had trouble moving the anchor around. I realized when grabbing & moving it, it looks like it's not moving against the time-line, but sliding the file behind as you go. Just odd to me.

 

Moving the anchor... OK, think about it like this... Imagine that you recorded yourself saying, "one, two three, four" starting at measure 17. If you open the sample editor you'll see that the anchor is positioned at the start of the word "one". And every time you hit play from measure 17 1 1 1 you hear the word "one" first. This is all stuff you'd expect to happen this way, but there's more to the "mechanics" of this than meets the eye...

 

The reason the region plays back at measure 17 is because the anchor point of the region is tied to an event that's generated at the point where you made the recording. And you can see this event by clicking on a blank area of the arrange page and opening the event editor. This is a special case where, instead of seeing notes in the event editor (like a MIDI region) you'll see instead the names of all the regions in your arrange page and the events which they're anchored to!

 

So the event at measure 17 is where the anchor point of your audio file plays from. Think about that for a second... The event is the point where the anchor point plays back from.

 

In this case the anchor is positioned at the beginning of the word "one". But... if you move the anchor point in the sample editor to the beginning of the word "two", when you play from 17 1 1 1 you'll hear the word "two". And if you start playback from the beat before 17, you'll hear "one".

 

So moving the anchor point within the audio file doesn't change the position of the event it's tied to. That event will always "trigger" playback of the audio file from its anchor point, wherever in the file it may be positioned.

 

More later, time permitting...

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