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JesseGMember
PICT1332.JPG looks like aliasing. That can’t be right though since it’s at full scale. Gosh I would hope not. 😛
For measuring modulation… http://www.MpxTool.com/ 8)
JesseGMember[quote author=”timmywa”]Alright, thanks for the info! Could you take a screenshot of your optimal settings. There are a lot of choices here! I appreciate it![/quote]
Just change the calibration to -23 LUFS, and see if it’s enough headroom. Everything else is fine on "R128-2" (which they are changing to R128-2011 to avoid confusion)JesseGMember0 LU per R128 should be perfectly fine. 🙂 ReplayGain calibration is where you want 0 LU to be at. -23dB LUFS is what R128 recommendation recommends (yes that’s repetitive).
Personally I don’t think they should be exposing most of these settings, unless you don’t have one of the EBU "Compliant" options selected. The fact it defaults to -18 LUFS doesn’t make me happy. That’s NOT compliant with R128.
JesseGMember[quote author=”mick heaphy”]so in theory i could use vac or use both to process 2 seperate sources through the same breakaway instence?[/quote]
you don’t need any more than 1 vac or 1 pipeline to do that… they both are multi-client on input and output on all of the APIs they support (wave/mme, ds/directsound, ks/kernel streaming)JesseGMembernew R128Gain, basically replay gain but using the new international industry standard in loudness measurement
i’ll check out your station tomorrow 🙂 cheers & always welcomed
JesseGMemberYes, it installs as its own driver separate from Breakaway Pipeline. You can use both at the same time as well, no problems. You can even uninstall Breakaway Pipeline manually in Device Manager without problem UNLESS of course you’re using Breakaway Audio Enhancer which will only accept input from one.
JesseGMember[quote author=”JesseG”]You should still be able to output the sound from SAM right?[/quote]
Did you try my suggestion?JesseGMemberYou should still be able to output the sound from SAM right?
JesseGMemberIt’s because your soundcard driver doesn’t support multiple clients on that "port". It’s probably Kernel Streaming (aka KS) which you’re using. Most soundcard drivers don’t support multi-client access on KS, so you’ll have to use DirectSound (aka DS) or MME (aka Wave) to do that, assuming your soundcard driver does support that. Most do.
The unfortunate side effect is that you’re usually going to need more buffering, so you get more latency. I recommend trying DS first, as it is usually more stable with lower latency than Wave.
JesseGMember[quote author=”irish”](1) what is the background of the people/person who developed this software.[/quote]
broadcast audio processing frrrrrreak[quote author=”irish”](2) What general settings would you suggest for someone listening to music or watching videos, with a realtek soundcard, and speakers that probably didn’t cost more than 2o euro euros, on a laptop and desktop.[/quote]
Whatever sounds best in the acoustic space they are in, from the preferred listening spot. It depends on what videos you’re watching too. If you’re watching movies and standardized TV productions, then you know pretty well what to expect of the input, and you can get by with a lot less processing than you would probably want on something like random YouTube videos.[quote author=”irish”](3) when playing something, after a while the program would make an odd noise, spluttering might be the best way to describe it. Then afterwards, i would hear crackling noise when playing something. I think though the format, of the file might have something to do with it, as it only seems to happen on some files, and then, not on all those type of files.[/quote]
It could be that you don’t have enough buffering or a "fluid" enough combination of length/count. Sometimes a combination that results in a longer total buffer can be less stable than some other combination. It also could be that you have a naughty driver on your system that has DPC latency spikes. Try disabling all networking and usb devices in your device manager, and see if that solves the problem. There’s also a few DPC latency checking programs around the internet, if you want to track down the culprit.JesseGMemberHey, stop the hatin’ guys. 🙂 He deserves as much respect as any of us, at least.
JesseGMemberThat isn’t possible without invasive memory programming, which would have to have crazy heuristics, and would still probably need to be adapted for a bunch of different versions, of just one software (of millions). And it would be a support nightmare. (im an ex game hacker, trust me, this is right up my alley)
Did you find the Breakaway toolbar in Windows’ taskbar yet? 🙂
JesseGMemberVery strange. No idea. Let us know if you guys do figure it out, and I’ll let you know if anything comes to mind suddenly, but don’t wait up for me to get home. That’s a weird one.
I won’t insult you with the obvious stuff like check what your default soundcard is set to be, etc. 😉
JesseGMemberIt doesn’t, but that’s a good idea for an upgrade. Thanks for the idea. Definitely could always use some more statistics.
That being said, in any testing I’ve seen… the tuner doesn’t rely on the pilot at all for RDS, which is why RDS still works when doing things like AutoPilot… Leif’s true-mono detection that fades out the pilot, for noise reduction purposes. 🙂 I’m not sure if that was part of the testing of the Omnia.9 @ NPR Labs, but I would guess it was, since it’s always on, and they would have noticed. Hehehe.
JesseGMemberNone of the current flavors of Breakaway input or output at 24bit.
I’m not sure if that’ll change in the future, but I do know that one of the things I finally convinced Leif to do was to allow the home user versions to input/output up to 192kHz (NOT with processing at that rate), so that we can use Leif’s ASRC, instead of Windows. Allowing 24bit shouldn’t be a problem. 🙂
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