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For FM there would be little to no audible benefit, on a real transmission, even with a reference grade tuner and playback gear. In 99.999% of actual listening situations, nah. No difference that anyone would ever notice.
But if you can output to analog from 24bit or 32bit, you still should use the highest quality you can, if there isn't something technically wrong/broken with a specific model/device. 🙂 I haven't heard of a soundcard sounding worse with increased bit depth, yet.
Swedish National Radio, one of the most respected FM networks for quality, is using a custom codec for their studio to transmitter links which runs at 160 kHz, 12 bit, mono. 🙂 Zero complaints about the sound quality, from anyone, or it wouldn't have even made it on air.
It doesn't mean you can't be slightly crazy about your sound quality, when it's free and easy. So yeah, crank it up! 8)
Now…. the input into a processor, that should always be as high a bit depth as you can get, and yes it does matter A LOT. The difference between a 16bit or 24bit input into a processor (from a 24bit studio path of course) can be audible on FM in some cases. On HD/Streaming/etc… it can be audible a lot of the time, even if it's only used for Billboard HOT 100. 😛
beej6, you're trying to use the same soundcard input/output with ASIO in two different software (BreakawayOne and something else) at the same time?
In general that's not possible.
[quote author=jadsawaya link=topic=5596.msg19665#msg19665 date=1543209647]station input sometimes has very high peaks (the Breakaway monitor is in the red). I searched old threads on this forum, and It seems many users were saying that adding an Attunuator helped with the distortion and cackling.[/quote]
If the distortion had already happened by the time BreakawayOne gets it from the soundcard, then no. That won't do anything at all to reduce distortion, because what's coming from the soundcard is already distorted.
What should be done is to re-calibrate the soundcard AND whatever is running into it, so that it's not clipping/distorting anything, and then ALSO given a bunch of headroom (space for the signal peaks to get higher) so that there's room left for mistakes to happen without clipping/distortion before it even gets to BreakawayOne.
Having peaks that are hitting -12 dB FS during normal operation (with music and DJs talking at the same time, or very dynamic music that has higher peaks at the same loudness) is a safe place to start. If it still ends up happening, you can calibrate it so the peaks are even lower. If it just keeps happening, then obviously… there needs to be some training for the gear (mixer/etc) they are using.
If you need a VST to just turn down the audio within the VST path, with a little bit more functionality…
https://www.sonalksis.com/freeg.html
Just curious, why do you need to turn down audio within the VST chain?
Spartacus is a VST plugin, but it doesn't have its own window/interface, and BreakawayOne doesn't currently have a default parameter UI of its own to control those VST plugins.
You are correct about it only being stereo. It will process Dolby Pro Logic just fine yes, since that is only using stereo channels. It will not do anything to alter the stereo/surround spaciousness, so any position information that can normally be extracted from Pro Logic will remain intact.
It will make the release speed up when there is more range (to increase gain) left in the band's AGC. The more range there is left, the faster it will speed it up. That control lets you set how much of that effect you want to have.
You should download the manual for the Omnia.9 if you want to know what some more of those do. 🙂 Most of them you will know what they are when reading the manual. Of course there any many things the Omnia.9 has, which BreakawayOne doesn't have, but it is still very useful.
There are m3u playlist links on the right side of the Icecast page, one for each mount/stream.
The mount point URL is the same as that playlist link, without the .m3u on the end.
Please uninstall ASIO4All as Milky suggested. It's not real ASIO, it just pretends to be ASIO and uses Kernel Streaming, which is already built into Breakaway products from day one.
So… to give you step by step instructions, we need to know how you want the audio to be routed…. from the point of your automation software or analog input, to the point where your streaming encoder is, if you're not going to use BreakawayOne's encoders.
You can draw it on paper and put a picture here if you want to, if that's easier. 🙂
The cheapest way to do all but one thing you wanted to do would be this:
The solution above is way more elegant, by taking away the ability to "mix B with the D so it can be heard [..] on [..] a new pipeline altogether". To do that, you would also need two Passthrough cores, and the switching on/off would be a slight hassle mainly because you would be able to hear both the Core 1 output through Core 2, and also directly, at the same time, if you have Core 1 also going directly into the headphones. It would also NOT be processing Core 1 through Core 2, which is what you described, which is why it would take a second Passthrough to handle it all correctly. This is true for ANY mixer/router that has the audio path you described.
I'm not sure why you want to process the output of Core 1 through Core 2 (double processing), but I figure you have a good reason.
I have an even nicer solution that'll allow you to have 3 volume knobs to easily control it as well…
This will also give you a fader you can select (in the top of BreakawayOne) for the amount of Core 1 output that you want mixed into the input of Core 2. Everything is the same as the flowchart above, except instead of having Core 1 going into Core 2, via Core 2 Input 2, you would have a "Common Monitor" (which has a speaker controller) with Core 1 HD Output patch point selected, and have the Common Monitor output to Pipeline C.
The only down side is that you'll have a little bit more delay since Common Monitor has to output to a soundcard, and come back in on a soundcard.
You would be best off trying to setup realtime recording to FLAC. You're going to lose more audio quality than anywhere else, by far, when encoding to mp3. Even if you're using the latest Lame at 320 with -q 0 etc…
Not all mp3 players support FLAC, even with alternate firmware (some do this way though, such as with RockBox).
There's multiple free software to record FLAC in realtime I think. Unfortunately not with the benefit of adaptive noise optimization though. Not fully lossless, but you can't hear it and it really does only even effect the samples that are too quiet to actually hear. Literally no sample you can even potentially hear is ever even touched at all with it, which is amazing. There's always lossyFLAC/lossyWAV. Monkey's Audio has an encoder that does this "hybrid mode" now, and it's pretty good compression at around 400kb/s for 44kHz stereo.
You could setup BaOne to have an HD path output to a virtual soundcard that you want to just record from, and then listen to it where-ever you want on an actual soundcard output with your common output.
Milky, what seems to be happening is that somewhere between their browser, and this web server, someone's blocking them from this site. It's not this site though. It wouldn't be an Error 403. It would be a normal HTTP response, and the forum would tell them that they are blocked (and we usually put a reason why, which is also shown).
So yeah, it's hard to say where it's happening algatelli. But it's not happening on this web server.
Yep unfortunately some drivers don't support kernel streaming but still have the API calls in the driver (which at best do "nothing" and then crash in the driver which means BSOD time). Usually it's an instant BSOD although. It's really extreme laziness that it ever happens, on the part of the driver developer/s.
Anyways, of course if there's no proper driver, the only option is to use another API (we really need to add WaveRT support) like ASIO if available (ASIO4All doesnt count its just KS), or get some hardware that does have a proper driver like you found. Hopefully with the kind of latency possible that you were looking for.
