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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.
I use an external USB3.0 spec sound device, with ASIO support. Have a look at the ESI U24XL https://www.esi-audio.com/products/u24xl/. This would probably function similar to the PCI Maya that you are familiar with.
MrKlorox is correct – buffers make a lot of difference. You need to change them in the M-audio control panel, and then use the test mode in BA1 to find the lowest latency. Sometimes, you can manually select a better option, but usually BA1 gets it right. If the buffer is too small, you will get "crackles" or artefacts whereas, if the buffers are large, it takes too long to fill/empty them.
Of course, because BA1 adds some processing to the audio, there may be nothing you can do except listen from the mixer, but I have been able to get the audio from music videos to accurately lip-sync through Breakaway 1.
