what sample rate do you use?

SoClose

Member
Oct 10, 2007
125
0
16
what sample rate do you prefer?

ive been using 48 but it seems allot of people are fine with 44. am i wasting my space with 48? granted if its for dvd then 48 is key but otherwise what?
 
There have been a few pretty in-depth threads about this recently; search and thou shalt find (though 44.1/24 bit is what was pretty much settled on)
 
I use 44.1/24 for a couple reasons. One is that the last time I used 96 my computer couldn't handle it, but it sounded nice. Two is at 96 my UAD-1 is cut in half. Three is the previous reason but for my converters, too.
 
44.1/24-bit here. I can't justify the extra storage/drives required for 88.2/96k and above.

However, for shits and giggles, I'm going to do an album at 48k next month. Just to see if it's any different HA!
 
44.1 here as well. For the hell of it, I did a few in 88.2 to see if there's a difference.... there "might" be, but it's pretty negligible. FWIW, SAW chewed up those 88.2 tracks like there's no tomorrow.
 
I actually don`t have a damn clue, but I checked out Internal mixing DVDs. I recall the dude sayin that you should use 44.1 for music thats going to be on cd. He also said that you should use 32-float because it actually is easier to your cpu. These are not my words but I`ve been using those parametres. Quality-wise I can`t really hear/tell the difference between 24 an 32 float.
 
I actually don`t have a damn clue, but I checked out Internal mixing DVDs. I recall the dude sayin that you should use 44.1 for music thats going to be on cd. He also said that you should use 32-float because it actually is easier to your cpu. These are not my words but I`ve been using those parametres. Quality-wise I can`t really hear/tell the difference between 24 an 32 float.

That's partly true, because most plugins work in an 32-bit enviroment. But since I bet your soundcard can only convert to 24 bit top, there is no benefit in sound to record with 32 bit. Just the file is bigger ;) But it can be true that 32 bit would use less cpu since the plugin can work with this signal easier.
 
11/8, bitches...

Jeff

Hahahahaha :D

I track at 48/24. If you use your HD space wisley, theres a bees dick of difference in file size between 44.1 and 48. granted theres little sound difference, but it makes me feel like i know what im doing :)

I master at 88.2/48. That way your files are bounced up to double red book standard and your plug ins can be more detailed.
 
yeah i am. how would that affect the tonal quality? your only increasing sample rate, how many pieces the waveform is cut up per second

Every process that you run your audio though degrades it. Analog audio is continuous. When you track at 44.1 your taking a sample rate from that continuous audio. When you upsample to 88.2 your taking a sample rate from a waveform that has already been captured. You have to process that audio, again. Then when your burn to disc it downsamples back to 44.1. Do you see what I'm getting at? Thats just my opinion tho. If its working for you then don't let me stop you.
 
Every process that you run your audio though degrades it. Analog audio is continuous. When you track at 44.1 your taking a sample rate from that continuous audio. When you upsample to 88.2 your taking a sample rate from a waveform that has already been captured. You have to process that audio, again. Then when your burn to disc it downsamples back to 44.1. Do you see what I'm getting at? Thats just my opinion tho. If its working for you then don't let me stop you.


ahh yes. i get you. i see where your coming from. in my ears, ive heard no difference when upsampling. but i dont burn to a cd straight from a 44.1k session, i bounce it down first :)
 
I'd think that upsampling by an integer ratio (like 2:1) wouldn't degrade it as it's trivial to introduce data that 'holds the place' necessary but doesn't actually change the actual waveform approximation... then again, I only study interpolation from a mathematical background, so I don't know if DAWs have a magical 'fuck the upsamplers' feature. Since you're already working with a discrete approximation of the waveform, and not moving it back to analog and then resampling it at the newer rate, I don't see where you're coming from.

For the record, it is trivial to prove that given any interpolation method one can add as much extraneous information after the fact as desired and not change a thing. This is what I would expect to happen, at least.

Jeff