So today I bought a focusrite Scarlett Solo And Used It In Cakewalk.I Had A Premade Drum Track So Plugged In To Record Guitar.So I Recorded The Guitar On Time But When I Stopped Recording The Track Was Off Time And I Don't Know Why
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Everytime I DI And Record My Recorded Track Is Always Off Time?
- Thread starter JohnOhofner
- Start date
Latency. It takes a few milliseconds for the recorded sounds to get converted to digital and saved to the track. What I tend to do is give a "1..2..3..4.." rake of the pick or thump on the strings against the drum track/click track before the recording starts, then zoom in and align those waves with the drum track visually. You CAN figure out the latency in milliseconds and program the software to offset your recorded audio by that much, but I am a knuckle dragger when it comes to such things.
*I use a Foscusrite 2i4 and Audacity or Reaper. Its not a platform issue, its just the nature of the beast. Thats why the focusrite stuff has the analog headphones outs, otherwise you can be trying to play in time as the D/A convertor is trying to out put the recorded sound AND process the incoming sounds. Ugh.
*I use a Foscusrite 2i4 and Audacity or Reaper. Its not a platform issue, its just the nature of the beast. Thats why the focusrite stuff has the analog headphones outs, otherwise you can be trying to play in time as the D/A convertor is trying to out put the recorded sound AND process the incoming sounds. Ugh.
USB audio interfaces are the worst for latency. You can't listen to your guitar (bass, vocal, etc) through the DAW while recording, while playing back pre-recorded tracks and expect them to be in time. A/D conversion takes a couple of milliseconds, USB transmission to the computer takes a few milliseconds, passing thru the DAW takes a couple of milliseconds, and then going back thru USB transmission to get back to the interface for D/A (so you can hear it through the headphone outs, or other monitoring outs) takes another few milliseconds. It all adds up. Analog mixers/tape machines never had this problem after 'sel-sync' was invented. Yay digital...
I own a Focusrite 18i8. Focusrite interfaces include a simple hardware digital mixer built into them for the exact purpose of allowing direct monitoring of incoming signals without the round-trip thru the DAW, including the ability to mix live signals from the inputs with output from the DAW, and feed that mix to monitoring outputs. This is what you're looking for.
Spend some time looking at this (access is installed as part of the driver software package) and you will come to realize its power.
Good luck.
I own a Focusrite 18i8. Focusrite interfaces include a simple hardware digital mixer built into them for the exact purpose of allowing direct monitoring of incoming signals without the round-trip thru the DAW, including the ability to mix live signals from the inputs with output from the DAW, and feed that mix to monitoring outputs. This is what you're looking for.
Spend some time looking at this (access is installed as part of the driver software package) and you will come to realize its power.
Good luck.
You can do a simple loopback test to calculate recording offset based on your current system and the sample rate of the session. All it requires is an output from your interface patched into an input and an audio pip. There are tons of guides online for how to do this easily with mathematically consistent results. Some DAWs include a tool that does it automatically for you and allows you to adjust the recording offset in samples or milliseconds.
Never had much problems with latency with my USB interface and DAW. But then, I use a custom built gaming computer with WIN 7 for all my video, photo, and audio editing. Chances are if you have latency, you're computer may have background apps and other software running and collecting data eating up processing power. To reduce any latency, check your computer's software and find what software is running at the same time you're recording. Find out how to shut down those background apps.
The latency is more from the interface than the computer. You can adjust the sample rate and buffer size as well and that can help. I have the same issue, I'm always off by 3/128th of a bar, whatever that is in ms I don't know, I just know that is where I have to adjust the starting point. It is noticeable. I'm on fairly old Echo Audiofires however and I suspect an upgrade to RME would lower that.
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