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You should consult with the manufacturer of the card, to find out if it’s capable of 192kHz sampling rate via ASIO. I would say yes, but it’s always best to find out for sure from the source before making a purchase.
On a side note…
http://marian.de/en/products/trace_alpha
That is cheaper than the card you’re referencing, and the output is DC straight, so you don’t need to do any low-frequency tilt calibration. Generally you’ll be able to squeeze a slight bit more modulation out of it without over-modulating. There still isn’t 64bit drivers for it yet, but the Seraph cards just got 64bit drivers a few weeks ago (!!!) so it’s only a matter of time now. Hopefully a short amount of time. hehe.[quote author=”didac”]Thanks JesseG, I think the soundcard can runs only at one clock frequency, then if you need use 192 kHz for Output, you need 192 kHz on input. It’s right?[/quote]
That’s how ASIO works, but you should be able to use Kernel Streaming (or Wave or DirectX) at different samplerates for input and output.
because ASIO opens the soundcard at whatever that maximum rate for input/output is, which is 96kHz. so you can’t use ASIO at 192kHz. I’m only repeating what was said, but it’s true.
@Tim, I would have to look into that before I replied. It’s likely that someone has already done the leg work on the statistics over at PLOUD.
@Bojcha, no that’s the currently public Breakaway Live input meter. ITU 1770-1 / LKFS.
The question is about input peaks, and I addressed it by saying that as long as the input loudness is relatively matching or otherwise not exceeding where the ITU input reference level (and matching indicator) are set to, that the sound will be ideal.
The peak signal levels don’t even tell us about how loud the peak is, much less the average loudness. For instance… Undo uses a psycho-acoustic model for its peak detection, not the signal levels, because it sounds quite wrong and screwed up going by just the signal levels "at face value". Leveling based on the signal peaks has no bearing on peak or average *subjective* loudness, which is also how the noise gating works in Breakaway. When you adjust the ITU input reference levels, it’s also adjusting the noise reduction thresholds. ITU’s 1770-1 aka LKFS is very close to subjective loudness. The mean deviation is a few tenths of a db father than ITU 1770-2 aka EBU R128 aka Loudness Unit aka LU… which I’m fairly sure will be in the next Breakaway.
Long story short… It’s best to set the average input loudness low enough to where you never have to worry about peak signal levels at all. And to make sure the ITU reference level roughly matches it so that the correct noise reduction is applied.
An average "not too slammed" modern recording:
Donald Fagen – IGY:
Still 6dB of headroom on IGY, and you’ll be very hard pressed to find anything much more dynamic than IGY when it’s at its peak loudness. That’s with an ITU reference at the new industry standard of -23dB (LKFS in this case). The ITU reference can be set up to 8dB lower than this., which would give you a very overkilled 14dB peak headroom on IGY. If you were a DJ, mixing in say… the latest Katy Perry song, at the same peak signal levels, the Katy Perry would blow IGY out of the water, ruining the mix, and potentially having a lot of background noise not being properly dealt with.
🙂
an average input loudness (the solid slow-moving part of the input meter) where the "reference ITU level" indicator (the line that doesn’t move unless you adjust the ITU ref level) is at on the input meters is best, and ideally setting it low enough so that you never have to worry about peaks. 🙂 the presets are designed with this in mind. cheers.
Ah the transmitter doesn’t even have composite input. Well there’s your problem. 😉 And yeah, get a schematic so you can bypass the audio front-end on it. 8) You won’t believe how much better it sounds using the composite from Breakaway.
Breakaway only synchronizes RDS if the RDS signal has the 19kHz pilot somewhere. The 57kHz (19kHz * 3) center of RDS is a suppressed carrier, meaning… it’s not actually there.
Coincidentally, I haven’t heard of a radio yet that actually requires the RDS to be synced with the 19kHz pilot. Someone from NPR Labs might say otherwise, because they have tested an UBER TON of radios.
If it handles BBP already without problems, and you have some CPU left, everything else you mentioned isn’t going to hit the CPU hard at all in comparison. Less machines = less things to break.
composite and rds are both mono signals. perhaps this is confusing you?
i’ll be honest, the way all of the posts are worded so far in this topic is confusing to me. i have to wonder if anyone else was confused also, and perhaps gave you advice based on the wrong idea of what you’re trying to say the problem is. hopefully i’m wrong, and you understand, that’s all that matters. hehe.
[quote author=”bennylein1985″]for the ogg stream i use 0.3db ATT
and for the he-aac i use 3.5db[/quote]
Try 2dB less drive into final clipper, use 25uS pre-emphasis (with de-emphasis of course), and see if you can bring your AAC gain up to -1.5dB. You should get a more easy to listen to and more dynamic sound overall, with about the same overshoot potential you have now.[quote author=”bennylein1985″]bbp adds pre-emphasis after agc, right?
i disable it still now….i use 15us before.[/quote]
The pre-emphasis is right before the clipper, so it’s after everything else except for the clipper.—
GRB (aka "feeling"), in my opinion… the stream link you just posted sounds the best of the ones posted yesterday/today I think, because it maintains ~all of the dynamics. 8)
Topic for comments/discussion of the links in "best sounding stream" topic. I made this to keep that other topic clean. 8)
That way we can keep the other topic down to 1 post per person, that everyone *should* be able to edit their own posts on, to update with new links etc, and then we can have 50 pages of discussion here. Also this will prevent links from getting buried etc, and give everyone a fair chance to be heard. 8)
I merged some posts into this topic, if you don’t notice above. hehe.
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By the way Benny, yes, that stream you have is pretty loud. Because of the way that codecs lose peak control, I think you can stay just as loud on your normal streaming codecs (like AAC) and sound a lot better because you’re not always clipping the peaks to a level that low, and removing that much detail, etc. It’s hard to listen to your stream for very long, because it is relentless.
AGC and Multiband both gate and freeze independently. (gating and freezing are two different things in Breakaway, and there’s some "blend" between the two also) Cheers.
