Utilizing the x1800 Sample Rate


Oh, from what I gather of that it isn’t so clear-cut, and simply comes down to what’s required for a certain scenario?


If you had a massive, massive amount of processing and could deal with a certain amount of lag/latency due to a kind of reverse direction feedback that has to be used (hence the delay), you can use many-point interpolative upsampling and FIR filters with windowing functions to either resample up or down to any distance and get amazing results regardless of scenarios/applications, but short of that, which is everywhere except off-line-processing with a powerful computer, there are going to be trade-offs. The x1700 can do a kind of cotton gloves handling of 44.1 by just staying in 44.1, but I still find 96 to be way way more musical on it. That’s just me, though. YMMV. The SC5000 sends out 96, anyway, so you might as well keep the x1800 in 96. If you have to use a lower sampling rate in a DAW, I’d recommend 48khz in the DAW if you can, as it’s an even division (and 44.1 usually puts the filter artifacts in the presence region of hearing). I’d also still record it to 96 and then downsample as a final step if necessary using something like iZotope’s stuff in post rather than just changing what the mixer is set as. That’s going to use more hard drive space, though, and if you’re playing from Traktor and the mixer has to be at a lower sampling rate for USB audio, then obviously that’s the deciding factor.


To keep it simple I have the prime sc5000’s hooked up to the x1800 and running serato pro 2.05. The majority of my music library is in .mp3 format. I have no control over how the mp3’s are made. I am simply wondering what the advantages of setting the mixer to 96 were. Since the mixer had the capability, i figured it must be there for a reason so why not give it a try.


I presume you’re having to use a Serato interface sending analog into the X1800? Yes, you’ll get higher performance out of the x1800 if it’s set to 96. The ADC stages, digital processing, and DAC stages will all perform with higher fidelity in 96.

The fidelity of the source music is moot to the benefit of the other links in the chain, as distortion and degradation is cumulative. The MP3 factor doesn’t present a brick wall situation or threshold where you might as well not use better quality stuff to handle it downstream. If the files were lossless, that’s much better, but MP3 doesn’t negate the benefits from higher fidelity elsewhere, rather it just lowers the sum total of fidelity, if you will, but only as one of those links in the chain, one of those variables subtracting from the sum total. Lots of other links in the chain and they don’t suddenly malfunction because of an MP3.

Now, don’t take this the wrong way, but it’s worth noting for completion’s sake that if you were playing 24/96 lossless files and there was no roll-off at all out to 48khz in actual recorded frequency, a lot of amplifiers and tweeters wouldn’t perform optimally. Hah hah. Weird, but true. That’s a luxury problem to have, though, dealing with actual ultrasonic content that makes it that far down the chain that you have to see if your actual sound system does well with it.


I am using serato, and using 1 sc5000 as primary usb and the other is plugged directly into the primary and they are in controller mode. Does that constitute an analog signal i thought that was digital?


All that matters in this case is how the sound is getting into the mixer.


With Serato sound does not go through the players.