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Welcome to the new forum, Plauri.
Actually, BA1 is very definitely a 64 bit application (on a 64 bit machine), but leif has noted in the past some complications with hosting a 64 bit VST within the environment, so has restricted them to 32 bits, within the 64 bit host.I’ll see if I can find some of the original comments.
Ah! Found this from Leif. His main concern is that the VST will crash and take out BA1 as well. By running a 32 bit “child” thread, only that thread can crash/
Here’s the original quote:-
“That would actually be quite difficult. I can’t run VSTs directly in the my process because then they could crash and take me with them, so I instead quarantine them in a child process. This is why 64-bit BaOne uses 32-bit VSTs by the way . This incurs delay though, and if they were to be inserted in the middle of the breakaway core, then we’d have to delay only the meters and patch points that precede them, or different parts of the GUI would be out of sync with the audio. It gets quite hairy to manage — I do in the Omnia.9 (and it was NOT easy) but there I am in control of the whole system, there isn’t a way for the user to add things.. adding unknowns in that location is a can of worms I don’t want to go near . What are these level-dependent VST plug-ins? VSTs have always been floating point in/out, meaning essentially infinite dynamic range, so I don’t know why that would be a thing — I certainly haven’t seen any.”Hi Mr K.
The new format is taking a little to get used to after using the other one for so long. There may very well be better or more efficient ways to arrange things. As we Admins figure it out, you may see some changes slide into place. We get the WordPress people to convert the old posts, and this is what they came up with, but nothing is set in concrete.What I am hoping we can provide is quicker responses to real questions, as I now have a closer alliance with the “people who know things”.
Hi Timmy. Nothing immediately springs to mind. Are you saying that the MP3s are clean outside of BA1? In another forum I Admin on, there are some recent comments about brand new Lenovo i7s inserting glitches in otherwise clean MP3 files, so maybe it's a Windows thing?
I finally caught up with Leif and can provide his response:-
AES192 means MPX output over AES at 192 KHz.
BreakawayOne cannot possibly provide AES because BreakawayOne is software. AES3 is a physical digital output format.
However, BreakawayOne provides digital MPX to the sound card. Everything BreakawayOne provides is digital — again, it's software. The sound card may in turn turn it into an analog signal, and then you have an analog MPX signal.If the sound card, rather than converting to analog, outputs a digital AES3 signal, then what you have is indeed AES192. But, BreakawayOne has no idea that that's what it's doing.
I have messaged Leif directly, but not yet received a response. Not unusual, but I will get back to you as soon as I hear from him.
The website mentions AES67, but that page hasn't been upgraded recently, so I suspect that AES192 may now be incorporated. I'll see if I can dig a little deeper.
Check your Junk Mail folder first, then email keith@claessonedwards.com
There is a "sticky" at the top of this thread about a new release to resolve BSOD in Win 10. Is this relevant?
There are all sorts of crazy algorithms to calculate the correct number of buffers. Basically, each "block" of samples must fit completely in the buffers allocated. It can't be split. BA1 attempts to find the optimum number of buffers, but sometimes, a little experimentation will yield a better result. The lower the number of buffers, the lower the latency, but, if you go too low, the samples won't fit into them, and artefacts will be the result. Getting the perfect fit of samples to buffers is not always possible, and I always allow a little head room just to make sure.
The video software I use has a slider where the video frame rate can be adjusted to match any audio delay, but it is just a lot easier to get the audio latency right.
This is what I use https://iconproaudio.com/product/utrack-pro/ and it is very fast.
I play a lot of music videos on a big projector screen, so lip sync can be a problem if there is any audio lag, but I can't detect any significant delay.I've never used Esi, but I can't imagine the behaviour you are describing as "normal" nor "acceptable". Are the drivers the latest for your PC? Is the USB port definitely 3.0? If you use an earlier USB port, it might work, but the bandwidth would not be supported over the slower speed.
The sample rate does not dictate the amount of delay, purely the resolution of the audio image, which is far superior than the ultimate sample rates you would be broadcasting. However, did you look at the other models in the Esi range on the link I provided? There may be a model that samples at the higher rate.
