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The "Jitter" is the change in timing compared to other "blocks" of audio date coming/going.
It can be above 100%, even without actual skips in the analog audio, because there's more than one block of buffering being I/O at a time.
It's still not recommended, but whatever works sometimes is what has to be done, no matter what the numbers say.
You can also hold your mouse over the slider/setting, and a yellow popup will tell you what the "backup setting" (original setting) is.
You can also compare two different presets, with the "Diff Preset" option in the load preset pages. This will set all of the "backup" settings in the currently loaded presets to the same settings that the preset loaded into "Recall Backup" has, and of course turn the sliders/buttons/etc text yellow if they have changed.
With AES and SPDIF both yes. There's no reason it shouldn't work completely fine with those adapters, if your AES/SPDIF soundcard/device that BreakawayOne is inputting/outputting from/to isn't breaking the standards. That obviously wouldn't be our problem, but just saying. 🙂
[quote author=Peter Tate link=topic=5616.msg19935#msg19935 date=1554562991]I found what it was…..[/quote]
What was it?[quote author=MrKlorox link=topic=5642.msg19933#msg19933 date=1554488181]I even bought a Plantronics Voyager BT 'headset' back in the day because I somehow got it in my mind that the VL tech went into it.[/quote]
Octiv "OctiMAX" inspired technology went into it, yes. I have a couple of those headsets. They are still great. 🙂Yeah, iTunes is probably set to use the wrong soundcard. It should be set to "Default" or to "Breakaway Pipeline 1". Cheers
[quote author=MrKlorox link=topic=5642.msg19780#msg19780 date=1550170227]Leif even made a broadcast TV processor[/quote]
Are you being super modest on purpose? Or… 🙂If it's "or" then, you might be surprised to know that Leif's audio processing effects around 85% of all broadcast television on Earth. Almost every box that Linear Acoustic makes is one of Leif's.
Not sure what the problem was, other than perhaps you may have entered the email address incorrectly?
There's multiple successfully activated new members today with Gmail addresses.
Thanks for mentioning it either way. I'll keep an eye out for a couple days to see if un-activated accounts using Gmail spikes.
1. The "ASIO4All" software only provides a virtual ASIO soundcard, which then accesses the real soundcard using Kernel Streaming. If you switch from Kernel Streaming to ASIO on the same soundcard using that, you will actually get MORE latency.
2. All versions of Breakaway support Kernel Streaming.
3. BreakawayOne is the only Breakaway you can try/buy now that has ASIO support.If you're not using ASIO4All or VB-Cable, I would uninstall those. I've had a few experiences with VB-Cable negatively effecting system latency.
The other issue you're having sounds like a combination of things.
1. Your Windows default playback soundcard should be "Breakaway Pipeline 1"
2. Your Windows default record soundcard should also be "Breakaway Pipeline 1" if you want to record what's being sent TO Breakaway to be processed. Hence, there will be no change from the audio that's being sent to it in your recordings.
3. You might not have spent any time adjusting (reducing) Breakaway's soundcard buffer size and buffer count settings. How much latency are we talking about? 40 milliseconds? An entire second?If you fix # 1 and 2, and work on # 3 for a little bit to find a good combination of buffer settings (4 settings total for input and output buffer size and buffer count NOT a big hassle to ~perfect), then you should be able to get the delay low enough to not be noticeable.
Final thoughts…
1. there is a bypass button in Breakaway (and in the Breakway taskbar toolbar), so once you get setup you can easily disable the audio processing, and hear the original.
2. you can switch the output soundcard in Breakaway any time you want, in the config, but the optimum buffer settings that work for one soundcard might not work stable (without audio glitching) for another soundcard. You should never have to reboot to change the output of Breakaway.There's a few things it could be. A VST plugin as Tim mentioned is one possibility.
Is it an analog input that you are processing? Or is the right channel pure digital silence?
Can you PM me a config backup file? You can create a backup file in BreakawayOne on the page
Home > Common > System > System Information
Click on "Backup (Save) Configuration Zip" button, and it will ask you where you want to save the .zip file.This way I can look into it much closer.
[quote author=James Augusto Nunes Da Silva link=topic=5513.msg19839#msg19839 date=1551555725]Stereo tool is a VST too so it's debatable here. He's hitting the max peak level inside the VST core from BreakawayOne. Reduce Stereo tool output so BreakawayOne VST core can have good Headroom to not clip anything. Declipper and Natural Dynamics add a lot of peaks (rly?) On audio so take care[/quote]
That is not correct information, Junior.
Steinberg's VST v2.4 plugin specification supports a theoretical dynamic range of 1680dB. Half of that is ABOVE 0dBFS. 🙂 In the real world, around 1000dB of dynamic range is common on most plugins that support "full" dynamic range.
BreakawayOne actual audio processing however "only" has 48dB of headroom above 0dBFS. Way more than enough for Declipper and Natural Dynamics to be run without a reduction in signal level. Not a great idea, but it still will be fine. In case you're wondering +48dBFS is 250 times as much signal as +0dBFS. 😉 It's a LOT of headroom.
I'm going to lock this topic, since the mono bass feature that is in BreakawayOne cores' "Misc > I/O > Proc" page has been fully addressed long ago.
[quote author=beej6 link=topic=5612.msg19724#msg19724 date=1548152496]Thanks Jesse G![/quote]
You're welcome 🙂 <3Kernel Streaming is 1:1 yes.
WaveRT writes sample data directly into the soundcard's hardware, so at that point it's really up to the soundcard to get it right lol.
WASAPI and WaveRT should be able to be supported. There's almost no reason to do WASAPI if you are using Kernel Streaming, since it's just emulating KS anyways, and you end up with the same latencies on average. (on plenty of systems i've seen worse latency performance from WASAPI in exclusive mode, better with KS).
WaveRT is way different though. That actually gives access to the soundcard's HARDWARE buffer memory, to the audio software that's using it, so that the audio data can be written directly into the hardware, without any need for ANY buffering in the audio API, other than what the soundcard itself normally uses for its own hardware's purposes. That's how it can sometimes even beat ASIO performance. 🙂
We shall see 🙂 For now, a soundcard with ASIO support is the way to go, if you need low latency for studio paths.
