Regarding the oft-related issue of sample rates, are you able to explain why Universal Audio upscales most of their UAD plugins to 192k for internal processing, then downscales back to the project rate.
There is no mention of it in marketing materials, and it is only mentioned in the Operation Manual in regard to latency considerations. Why would they do it if it was not necessary, especially as the extra processing undermines their UAD cards’ plugin number capacity and latency?
And what implications does it have for DAW users, who do a lot of complex processing to get their end result which may be just 44.1k, or even an MP3?
And just to be clear, this is a separate issue from what people can distinguish between as far as the end product goes, where the testing has involved only a single sample rate change and has shown people generally cannot distinguish.
192k can be beneficial for many kinds of non-linear processing - distortion modelling, compression etc. Having this up/down conversion builtin guarantees the quality of the conversion processes (to UADs standards), while leaving the engineer free to use whatever rate they want in the project.
Gone through the whole video and I almost bought it.
Fact is that more then 20 years ago the CD format was very often discussed and it became very clear that all CD players incorporated rounding error algoritmics to make playback of CD’s sound good (this is in fact told by software engineers who build these cd players), so in my book there is Always rounding algorithmics involved, the steps that have to be rounded become smaller as the samplerate increases.
What iBM said… sample rate is a totally different topic. (but not without it’s own myths, upon which mentioning would likely throw this whole thread off track.)
If your A/D convertor in your interface is 24bit and as such can only deliver 24bit digital data, what is the use of padding the signal with nothing when storing it in your computer? Cubase already works internally on 32bit float, regardless of what the bit depth of your recordings is.
Do any of you know if there are interfaces available that convert at 32bit?
Jan
If anyone has to work at different sample / bit rates just try them out first to see if there are no bugs at that level. Seems to me that a fair amount of users of the higher 96 or 192 get a fair amount of problems that other users don’t experience.
32 bit audio benefits. Don’t think there’s much in it but I do set Cubase at that + 44kHz. If your ears can hear it then use it if it sounds better. Psychologically my head secretly thinks that using 32 over 16 means I get less bugs. But that’s just me and my monkey.
32 bit floating adds NO more bits to 24 bit integers. In fact, below 0db, there is NO practical difference.
The ONLY advantage of 32 bit floating is that it is impervious to overloads, which helps DAWs cope with a wider range of users as gain staging is not as critical.
Have to thank all the experts who provided information here… I was not aware that I should turn off Hyperthreading… not the subject of the original post, but highly useful ( and mentioned in Steinberg Knowledge Base). Helped a lot with my performance issues with Studio Drummer, my go-to drum set.
Also, love those videos from Xilph - Monty Montgomery - very well explained.
If i creat a empty project with 32 floating bit depth setup and import 24 bit audios for mixing. What will happen, will the 24 bit audios be converted to 32 bit format?
IF not, does it mean the extra depth is used as a virtual headroom?
What happened a 24 bit depth audio is imported to a 32 float project for mixing? Will the audio be converted to 32 float F
format? Is the extra bit depth used to add extra headroom to emulate the analog gears.