Latency ??'s

fuzzsniffvoyage

Well-known member
I am currently using a lap top i7 8 GB Ram, Tascam US-1800 interface, Sonar X1 Producer/expanded DAW. Any way, for the longest time I was using my settings at 16 bit/44k sampling rate, with no problems what so ever. Then I was reading that maybe I should be using a setting of 24 bit/ 96k sampling rate, which caused, what seems to me massive latency problems. I tried changing my buffer setting, which did not seem to help. I was getting a lot of popping, crackling and stoppage while recording and play back. Also when doing over dubs, when I listen back, every thing is out of sync, like my 2nd pass is late like a 1/2 sec or so. And this is happening with no plug ins, so my cpu isn't being taxed to hard.

So I changed my setting to 24 bit/ 44k sampling rate, and my computer seems to like this setting and no more latency problems.

I guess my ?'s is does it really matter that much the sampling rate?

I read that the 24 bit will give me more head room, correct?
 
I am monitoring through the Tascam's headphone jack while tracking and through the monitors while listening to play back.

So, I am not monitoring out of the computer itself or the CPU???
 
96K is way overkill since CDs are 44.1K and MP3s are much lower. Not to mention the kid who is buying music already lost half of his hearing from his ipod! 44.1K or 48K / 24bit is fine.

BUT now there's the "mastering for itunes" thing where Apple recommends 96k/24bit or higher before mastering. But Apple says a song will benefit as long as its at least 48k. This is a new technology hardly used by anyone, but it's out there and must be mentioned. If you mix at 44.1k you can't use it....
 
You have two issues. One is the sampling frequency. For whatever reason your setup isn't working with 96k, but that's okay as it is more than necessary. I like 48kHz. It's video/film compatible and slightly "better" than CD. I don't mind downsampling for CD and having some future proofing.

The other thing is that the US-1800 has its own input monitoring feature, crude as it is. It appears to simply sum all the inputs and mix that with the playback from your DAW. You use the Mix knob to balance the playback from the computer with the live playing at the inputs. That will give you zero latency monitoring of your inputs at the expense of the ability to adjust the mix separate from the record levels.
 
I am currently using a lap top i7 8 GB Ram, Tascam US-1800 interface, Sonar X1 Producer/expanded DAW. Any way, for the longest time I was using my settings at 16 bit/44k sampling rate, with no problems what so ever. Then I was reading that maybe I should be using a setting of 24 bit/ 96k sampling rate, which caused, what seems to me massive latency problems. I tried changing my buffer setting, which did not seem to help. I was getting a lot of popping, crackling and stoppage while recording and play back. Also when doing over dubs, when I listen back, every thing is out of sync, like my 2nd pass is late like a 1/2 sec or so. And this is happening with no plug ins, so my cpu isn't being taxed to hard.

So I changed my setting to 24 bit/ 44k sampling rate, and my computer seems to like this setting and no more latency problems.

I guess my ?'s is does it really matter that much the sampling rate?

I read that the 24 bit will give me more head room, correct?


Out-of-sync audio is not caused by latency, per se. It is an indication that the latency compensation in either your DAW or your drivers is massively broken. That's a bug.
 
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