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My old pc with breakawy asio and a esi maya juli@ is almost end of life.
I had the soundcard with asio with no delay at all. And i want that on my new pc also.
Because i use breakaway for a radioshow for my own sound on my headphone with absolute no delay.
So..what do i need ?
1. wich soundcard ? ( because new pc doesn't have the old pci but only pci-e )
2. breakaway one..but what for core ?I don't know sound cards, but you can get upgraded to the Full FM core for free since you own Breakaway ASIO: https://www.claessonedwards.com/free-upgrade/
However if you're not planning on using the FM features, you might be able to get a fully decked out HD core and the Management Package. It all depends on what you need to do.
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.
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.
Oof, my new ESI U24XL must be defective. It doesn't even go above 48kHz. It straight up refuses to even hold 48 and always downshifts to 44.1 without resampling. It just skips frames and sounds terrible.
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.
[quote author=Milky link=topic=5935.msg20412#msg20412 date=1581110739]
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.
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Definitely using latest drivers and a USB3 bus. Gave it a second chance earlier and it was just as disappointing.I'm sending it back and ordered a MiniDSP USBStreamer box (https://www.minidsp.com/products/usb-audio-interface/usbstreamer-box). It's mostly a single use device, but it's the only other ASIO interface with SPDIF I could find for my price range. Luckily it should be all I need to run my video game console audio through BreakawayOne at low latency, which was my primary interest.
Sorry to hijack the thread :-[
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.Here is some input from Leif:
"Sample rate has very little to do with latency, actually. The important part is the buffer size.
A sound device running at 192 kHz at buffer size 256 will have similar latency to a device 48 kHz at buffer size 64.When I last tested the Behringer UMC202HD, latency was acceptable to me, at least for a USB device.
However, I recommend using separate devices for low latency monitor out (which BreakawayOne Full FM can do) and MPX out.
The Behringer (at 48k) should do great for main input and monitor output. For MPX out, I recommend the CM6631A USB dac, or Sanskrit 6th gen USB dac.
Best,
///Leif"Thanks Keith and Leif! So how does Processor Latency Mode affect the buffer? [Core> Misc> Processor> Latency Mode] Not talking about the separate LL module.
I notice if I set my new ASIO SPDIF interface to a buffer size of 32 at 48khz, I can only use High Quality in my HD core without noise artifacts. However a size of 64 at 48khz can work fine in Extra Low Latency, but still has audible artifacts when I choose the standard Low Latency mode. I'm guessing it's supposed to be buffer size 1x @ Extra Low Latency, 2x @ Low Latency, and 4x @ High Quality?
Still, it probably shouldn't be making any buffer related noise at standard LL, but not when on Extra LL, right?
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.
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