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Old 06-26-2014, 12:49 PM   #1
StuartJ
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Default Best initial set up and 88.2KHz

Hi There!

I'm new to Reaper and want to know the best way to configure it for a Mac. I'm having to import a bunch of tracks recorded in another DAW. They were recorded at 88.2KHz. Will Reaper handle 88.2KHz?

Is there anything else I should know about the initial set up to help make things run smoothly?

Thanks for all your help.

StuartJ
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Old 06-26-2014, 04:45 PM   #2
serr
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88.2k runs every bit as well as 96k in Reaper. I've just started doing everything in 96k as that has become the standard release format but I've mixed a number of large-ish 200+ track sessions in 88.2k and even for 5.1 surround.


Reaper comes out of the box ready to go for live sound mixing (or so it appeared to me). Which of course is impressive! Editing features are just under the surface. Check out the Action menu.


As for settings...
Set your disc buffer low to achieve 11ms or less latency for running live sound or live performance. Or to overdub if for some reason your interface doesn't have a cuemix/monitor feature.
Set it to 1024 and forget it for post work (where there are no latency considerations).

That's all SOP for any DAW.

Here's one:
In Preferences/Audio/Buffering set audio reading/processing threads to 1. It is 'auto' by default and may set itself to the number of cpu cores you have otherwise. This is 1 by default for recording/live sound mode but defaults to auto for playback until you change it. Which can lead to clicks and pops if you start working the system even at low cpu use.


That's about it.
Reaper is so stable in OSX that you can run live sound with it while recording all the inputs to multitrack. Studio post work seems trivial to Reaper no matter what you dial up.
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Old 06-28-2014, 08:39 AM   #3
StuartJ
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Default Best initial set up and 88.2KHz

I have a couple questions about your preference suggestions and a couple initial problems you might be able to help me with.

1) In the preferences should the buffer for recording be 11ms or 1100ms? I'm thinking 11ms.

2) For playback it should be 1024ms. Do I need to quit and restart reaper for these changes to take effect?

3) I've imported files from another DAW and the timing is slightly off on them. Any ideas why this might be?

4) During playback the the box with the minutes/seconds flashes red and then sometimes I lose audio. Any idea why this might be happening? I think that the preference setting don't go into effect until after a restart.

Any and all help would be greatly appreciated.

Thanks,
Stuart
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Old 06-28-2014, 08:40 AM   #4
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Oh, I see the "Apply" button in the prefs, so I assume a restart isn't needed. Any ideas about the other problems I'm having?
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Old 06-28-2014, 08:56 AM   #5
serr
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The disc buffer is in Preferences/Audio/Device page.

Check the box next to 'request block size' and enter the value. This value is in samples. NOT ms.

Again, set low (128 samples might be a starting point to try) if you need to run live sound or for live performance where you need to monitor the live sound through Reaper in "real time" (usually agreed to be 11ms or less total system latency). Look up running loopback tests to help dial this in.

If you are only mixing, there's no need for "real time" monitoring of inputs (as you aren't using any - you're playing already recorded audio from your hard drive). For this, just set the buffer to 1024 samples. No need to make the system work for a feature you aren't using.


There is also a media buffer in Preferences/Audio/Buffering. This one is in ms. This is only to buffer media for playback. This has no effect over recording or system latency.

#3 - Check that you didn't accidentally change the playback rate slider in the transport dialog box. Should be 1.0

Also check that you are running your system at the correct sample rate. Preferences/Audio/Device checkbox for 'request sample rate'. Reaper gives you the ability to run your hardware at different sample rates from the project setting (not normally what you'd want to do).

#4 - Probably because your system can't handle the 11ms latency you were trying to set. (Live sound settings for "real time" latency do require some premium components - especially a SSD for system and audio). Again, set your disc buffer to 1024 samples.

Last edited by serr; 06-28-2014 at 09:04 AM.
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Old 06-28-2014, 12:47 PM   #6
StuartJ
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Thanks for your help. I'm still having playback issues and I only have 7 tracks loaded.

I'm attaching some screen shots that might make it easier for you to see where I may need to change a setting.

Any idea why the tracks are rhythmically not aligned once imported?

Also, how can I set the grid to 16th notes instead of 8th notes?

Sorry to have so many questions. There is a bit of a learning curve. :-(

Thanks,
Stuart
Attached Images
File Type: jpg Reaper-Audio-Device-settings.jpg (39.9 KB, 322 views)
File Type: jpg Audio-Buffering-settings.jpg (64.2 KB, 340 views)
File Type: jpg Audio-playback-settings.jpg (54.1 KB, 325 views)
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Old 06-28-2014, 12:53 PM   #7
serr
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#1 Check the box next to 'request sample rate'.
Set the block size (your disk buffer) to 1024. 2048 is usually not supported.

#2 #3 look ok.

Make sure the project sample rate (project settings under the Reaper File menu) is also set to 88.2k
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Old 06-29-2014, 08:45 AM   #8
StuartJ
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Thanks again, serr, for your help!

Well, I made your suggested changes and the software/computer still can't seem to keep up and I only have 7 tracks and no plug-ins running. What happens is the transport box (to the right of the controls with the song time in it) flashes "red" and the sound goes in and out. :-(

Here's my configuration:
Mac OS 10.6.8
2 GHz Intel Core 2 Duo
4 GB of 667 MHz RAM

This should be enough processing power, shouldn't it?

Any and all help from anyone about this would be greatly appreciated! :-)

Sincerely,
Stuart
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Old 06-29-2014, 09:17 AM   #9
StuartJ
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BTW, I'm running the 64-bit version. Maybe I should try the 32-bit one?

Thanks,
Stuart
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Old 06-29-2014, 09:53 AM   #10
serr
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Quote:
Originally Posted by StuartJ View Post
Thanks again, serr, for your help!

Well, I made your suggested changes and the software/computer still can't seem to keep up and I only have 7 tracks and no plug-ins running. What happens is the transport box (to the right of the controls with the song time in it) flashes "red" and the sound goes in and out. :-(

Here's my configuration:
Mac OS 10.6.8
2 GHz Intel Core 2 Duo
4 GB of 667 MHz RAM

This should be enough processing power, shouldn't it?

Any and all help from anyone about this would be greatly appreciated! :-)

Sincerely,
Stuart
What hard drive(s) are you using?

It's definitely plenty of processing power. But your hard drive system needs to be up to speed as well. (That's the next guess since you haven't mentioned what you're running for drives yet.)

For example, a stock 5400rpm HDD with OS/apps/audio would absolutely be expected to struggle with audio work.

What does Activity Monitor show when you get the errors?

Quote:
Originally Posted by StuartJ View Post
BTW, I'm running the 64-bit version. Maybe I should try the 32-bit one?

Thanks,
Stuart

This depends entirely on the plugins you are running. Use 32 bit Reaper with 32 bit plugins. 64 bit for 64 bit. If you have a mix of both 32 bit and 64 bit plugins, then you get into running 64 bit Reaper and bridging any 32 bit plugins to use them.

The stock Rea-plugins will obviously be correct. It's all about your 3rd party plugin collection.

****

The interface connection is the next thing to investigate.
Does that unit have it's own control panel (for setting clock source, etc, etc)?
Everything set up correctly?


Further:
You can find out if your trouble is with the interface/processing vs. the hard drive. You can test the interface and system and leave the hard drive out of the equation by running live sound through the system. (ie. No audio streaming from or to the drive.)

Assign a track to a mic input.
Live sound 101: Start with that channel fader all the way down!
Record arm the track. Monitor mode auto or input.
Track output to master.
Master assigned to an output pair on the interface and patched to your monitor system (speakers).

Plug in a mic. Amplify your voice through the system. (Careful inching that channels volume control up so you don't blast yourself with feedback if this is new to you!)

You should hear a very noticeable latency delay with a 1024 sample buffer.
Now set the buffer to 128 samples. The latency should be gone or almost gone now.

Alright... Is it working?
Got live sound going through the system with no distortion, clicks, or pops?

If yes, you've just verified your interface and system are working.

Last edited by serr; 06-29-2014 at 10:15 AM.
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Old 06-29-2014, 09:58 AM   #11
StuartJ
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OK, the 32-bit behaved the same way. BUT, I think I may have solved this in the 64-bit version. I checked the "Auto-detect the number of needed audio processing threads" box in Preferences/Audio/Buffering.

Serr, I think you recommended this be set to 1, but setting it to auto-detect seemed to solve it for me.

However, I still have alignment issues. The tracks start out sync'd nicely, but over time they fall out of sync. Any idea on what might be causing this or how to fix it (other than manually)?

Also, I'd really like to have the grid be set to 16th notes rather than 8th notes so I can verify the timing of all the tracks. Does anyone know how to do this?

Many thanks,
Stuart
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Old 06-29-2014, 10:03 AM   #12
fwd0120
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Maybe you are having disk IO problems. Make sure the audio is on your HDD, not a USB stick (I made this mistake in a session once, wasted too much time). With waveout in the audio preferences, try 8X1024. Your specs are kinda low, but I used comparable specs for several years and no probs.
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Old 06-29-2014, 10:05 AM   #13
fwd0120
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Also, have you tried resetting the reaper config?
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Old 06-29-2014, 10:19 AM   #14
serr
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Quote:
Originally Posted by fwd0120 View Post
Maybe you are having disk IO problems. Make sure the audio is on your HDD, not a USB stick (I made this mistake in a session once, wasted too much time). With waveout in the audio preferences, try 8X1024. Your specs are kinda low, but I used comparable specs for several years and no probs.
Put a SSD in that machine for OS/apps/'high performance audio space' and that would be a plenty powerful system. You get far more bang for the buck with that $120 upgrade than anything else you could do. (250GB SSD's are currently going for around $120. $90 if you catch Tiger Direct on the right day.)

For DAW work, the part of the system that requires the highest performance is the hard drive.

Last edited by serr; 06-29-2014 at 10:39 AM.
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Old 06-29-2014, 10:28 AM   #15
serr
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Quote:
Originally Posted by StuartJ View Post
OK, the 32-bit behaved the same way. BUT, I think I may have solved this in the 64-bit version. I checked the "Auto-detect the number of needed audio processing threads" box in Preferences/Audio/Buffering.

Serr, I think you recommended this be set to 1, but setting it to auto-detect seemed to solve it for me.
OK. Well, that's counter to what I've found but there are a number of variables at play. It's even possible that it's fallout from coaxing this obsolete UAD pci card plugin system to work in my system too. *

Quote:
Originally Posted by StuartJ View Post
However, I still have alignment issues. The tracks start out sync'd nicely, but over time they fall out of sync. Any idea on what might be causing this or how to fix it (other than manually)?
Out of sync with what? Are you saying that audio tracks that were recorded together now drift from one another?

Look to see that you haven't inadvertently time stretched one of the items. Or accidentally moved the play rate slider in the transport window to something other than 1.0 and then proceeded to record more audio.

Quote:
Originally Posted by StuartJ View Post
Also, I'd really like to have the grid be set to 16th notes rather than 8th notes so I can verify the timing of all the tracks. Does anyone know how to do this?

Many thanks,
Stuart
You'll have to look up the grid features in the manual. It's very rare that I use it. I'll snap a grid to my audio if I really have a ton of bar/beat arrangement to do (indecisive client), otherwise it's much quicker to just edit the audio freehand (weather or not a click was recorded to).

* The only thing I've found that even affects my system performance seems to be these old UAD plugins. With no UAD plugins, ANY settings for audio threads work. Like it has no effect. Start pulling UAD plugs up and now I can 'break' the system with 'incorrect' settings (even though there is only maybe 7% cpu use and 800mb ram used). I arrived at 1 audio thread as the 'best correct' setting. I remember reading a number of forum posts also recommending 1 audio thread. That's my story.

Last edited by serr; 06-29-2014 at 10:36 AM.
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Old 06-30-2014, 07:06 AM   #16
StuartJ
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Thanks for everyone's help.

I reimported the tracks and they seem to aligned properly with no timing issues.

I found where I was able to change the grid to 16th notes. Stumbled on it under Options/snap grid/snap grid settings.

This seems like a very robust program with lots under the hood. I don't understand most of it! I'm sure I'll have more questions going forward, but thanks to your help, I'm at least up and running. I'll try and do some tracking today.

Thanks,
Stuart
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