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Outbound connection is for the remote control software. It was added so that remote controls can work where even NAT traversal is normally difficult/impossible, like carrier class internet ala… a cellular hotspot, etc… or if you just don't have any control over port forwarding behind NAT.
https://en.wikipedia.org/wiki/NAT_traversal
This will try (every 10 seconds) to connect to a remote control that is setup in listening mode on a specific port number. This does mean that the remote control that's listening does have to have that port forwarded to it, if there is NAT.
Another thing that can effect this is the "Power" control on the "Multiband" page of Audio Processing. The lower it is from Infinity:1, the less change there will be between the input and output of the multiband. Set on 2:1 there will only be a 50% change from the input audio, to what the output can be. This is intentional on a number of presets, and is part of their sound, but please do play with that setting also, to see how it effects the sound (and you may notice the processing meters changing too)
If it's a little glitch every 10 seconds, then it's some other software or driver/hardware that is running on the computer that is causing it.
You should check out this software, and see what information you learn from it. 🙂
https://www.resplendence.com/latencymon[quote author=warmeman link=topic=5732.msg20002#msg20002 date=1555933131]
Please explane..my old breakaway with esi maya44 has no delay at 192khz asio, but with the esi maya44 xte at 96khz asio there is a delay. Only different is the lower samplerate. Please explane..
[/quote]If you are lowering the sample rate, but not lowering the amount of samples used to buffer, then you are increasing the TIME that it takes for audio to go through that buffer (aka latency). It is not ONLY because of the sample rate.
Just like if you increased the amount of samples used to buffer, but did not change the sample rate, the TIME (latency) would still increase even though sample rate did not change.
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You should be able to use less samples/size in your buffer/s, if the sample rate is lower, because the same amount of samples takes more TIME for audio to go through it with a lower sample rate. Understand? 🙂
Think of the buffer samples number like the amount of cars on a section of road. Think of the sample rate like the speed limit. =)
[quote author=MrKlorox link=topic=5733.msg19982#msg19982 date=1555386271]It seems intentionally slightly random so you can't schedule your own interruptions.[/quote]
Correct[quote author=MrKlorox link=topic=5733.msg19982#msg19982 date=1555386271]I recommend downloading the trial on another pc and timing it between the first two nags after launch a few times. Remember it needs to be processing audio for the internal timer to start counting, that way it doesn't interrupt silence.[/quote]
This is the exact reason that people BUY backup audio processors for their radio stations. You're recommending that they find a way to steal one of the two processors they would need to do this. Thanks buddy.But yeah, it won't work anyways. 😉
Technically there's no way to do this is you're updating the final component before the listeners. Luckily BreakawayOne is NOT the last component. The streaming server is. So really what should be done is for there to be some backup audio that plays on the server if the source is lost/dropped.
I recommend checking out something like this
https://www.liquidsoap.info/
if you want some real power/features to do the things you want to with your streams, from the server side, when you need to do things like update BreakawayOne, or if your internet connection/s go down, etc.It could also be that the Roland itself just isn't capable of "low enough" latency for you, no matter what ASIO software/settings you use.
Have you ever used an RME soundcard? :))
[quote author=dancemusicradio link=topic=5639.msg20085#msg20085 date=1557329848]
Hi there. I haven't looked into this any further myself as i have no need to use itunes with breakaway pipeline. It was curiosity to see if it could work that caused me to have a look to see if it would work.
[/quote]All good. 🙂 If it's a non-isolated problem, I'm sure you won't be the last to mention it.
Broadcast audio processing, vs studio audio processing, is really the big difference that you're experiencing in trying to do that.
Broadcasters have the advantage of over 60 years of innovations in audio processing that the recording industry has almost completely ignored, while they stayed largely complacent.
One good example is…
1956 Gates STA-Level, the first compressor that has a "gate" to stop the compressor from adding additional gain when the input levels fall below a certain threshold. A "gated compressor". This innovation alone is something that is still to this day a BIG deal for a compressor to have, and almost completely non-existent even in the entire history of recording industry's master recordings. Finally within the last 10 years that compressor gained a nice cult following in the recording industry, and even appeared on a Jay-Z album cover a few years ago. Unfortunately not even that innovation itself has captured much of any interest from the recording industry.
On the other hand, they have eventually caught on to 1-2 other big innovations in broadcasting, such as…
1972ish Durrough DAP-310, the first multiband compressor. 3 bands mono box. This is an example of a very big innovation in broadcast audio processing that eventually (in the era of software plugins) finally did catch on. One of very few examples.
Unfortunately there's a thousand examples of the recording industry ignoring innovations, per example of them adopting another. Half of them are still semi-secret as well. Most broadcast audio processors have at least 2-3 secrets in em, some have over a hundred. "Special sauce" 🙂
What happens if you only install iTunes, and not Breakaway Audio Enhancer? Just wondering.
Sounds like iTunes might be fiddling with the Windows audio subsystem, even when it's not running. Lovely. Insert Louis Rossmann quip here.
If Breakaway Pipeline doesn't show up in iTunes soundcard output preferences (I'm assuming there is such a preferences because you are telling me, i do not use Apple products)… then Apple are specifically doing this on purpose, and you need to contact Apple support to have them fix their problem that they created. Good luck with that. 😀
If we see a bunch more reports of this, obviously we'll communicate with them, I still have some decent contacts there (including a vice president) so… I can make things happen. But obviously I'm waiting till we get flooded with people reporting the same problem before I bother anyone at Apple about it.
I hope you can figure out the problem. 🙂 Have you tried re-installing Breakaway Pipeline? reinstalling iTunes? reinstalling QuickTime or whatever other Apple software that iTunes or any other Apple product/s that you have installed on your computer?
If you're playing 48kHz audio in Windows….. into a soundcard that's operating at 44kHz, then it's Windows which is going to do the sample rate conversion. This is completely independent of any software/hardware using the soundcard input/output.
In Breakaway Audio Enhancer, the adaptive sample rate conversion is being done on the input side. The processing core will run at 44kHz or 48kHz depending on the selected OUTPUT sample rate. If you're outputting 48kHz from BAE into Wave or DirectSound, and you have Windows set to default that specific soundcard output to 44kHz, then Windows will do another sample rate conversion so that it matches.
So basically, you need to make sure that not only Breakaway is using the correct settings, but that Windows settings for the soundcard input (record), and output (playback) match the sample rates you're trying to use.
With one exception…. if you're using Kernel Streaming ("KS") to access the soundcard/s, it will over-ride and also bypass all of those parts of the Windows audio subsystems, and force the sample rates, bit depths, etc.
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I agree that there's better solutions to your problem than altering the media library itself.
There's some pretty good sample rate conversion in some video players now that sound a lot better than Windows. Also, not all WIndows are created equal. Sample rate conversion in Windows was quite bad until Windows 7 Service Pack 2 (actually it was this hotpatch which fixed the problem originally), and it's been improving since then. It's still not as good as BAE's, so you may hear an improvement by using one vs the other. You certainly don't want both though. 🙂
That aside…. 🙂 were you able to find a sound closer to what you want? I see you disabled bass enhancer in whatever preset you used as a starting point.
Did you find the controls for Gain Riding -> Speed? That's the overall speed of the Input AGC.
The authorization servers are now offline, after having removed the last product that was being sold that needed it, which was Breakaway Audio Enhancer, and giving time for those last people that bought it to authorize it to the computer they will use it with.
Sorry that it was done quietly but, yes, the old authorization server that handled all of Leif's software until the last few years. Upgrade paths were provided for its Breakaway users to switch to BreakawayOne for free.
The old authorization server was also setup to stop giving out new authorizations after 5 per license anyways. We figured this was good enough for people to be able to correctly register Breakaway on the 1 computer they are allowed to put each single license on, and to deal with people's OS re-installs etc. And that it was low enough to prevent any real abuse for whatever reason, with minimal hands-on support needed for legit outliers.
So if you were able to "re install on fresh hardware" in the past, then you were being a bit naughty already. 😉
The new authorization server/methods will lock the license to a specific hardware, the first time that it is authorized. It can be re-authorized as many times as you want, if it's the same hardware.
[quote author=timmywa link=topic=5618.msg19746#msg19746 date=1548944984]I don't think that controls levels into the encoder(s)[/quote]
[quote author=aleph99 link=topic=5618.msg19723#msg19723 date=1548129799]where do I control the listening volume?[/quote]
Listening volume, not streaming volume, so nick_ca is correct.
Have you found even one single VST 3 plugin that you want to use which doesn't have a VST 2 version also?
Not saying it couldn't be supported in the future, but… seems to me that everything is already available as VST 2 plugins, and so why are we trying to confuse things by adding support for something that isn't necessary (at the moment), and wouldn't benefit the sound quality either?
