Posts by piotrmaj

    Yes, noticed this as well. Stability dropped, there are tons of memory leaks plaguing this application, and it is probably healthier to restart it from time to time. However you don't have to hard restart the Mac to recover. Right click on the RM icon in the dock, then hold "Option" and choose: "Force quit". In one case I had to unplug USB cable and plug it back to fully recover. On the plus side: Performance management started to work like it should. :)

    There was one hardware revision recently, which added SPDiF slave mode, but there is no official information about any other components being updated. I have 5 years old toaster and yes, compared to stage it is 3x slower. I think of it "vintage vibe" :-)

    yeah, it is broken for many people: see this thread for example. I filed a few tickets as well (included video showing the problem), last answer from support was: "we haven't been able to reproduce this so far. It doesn't seem to be a very common issue but we've filed a bug report for it." Which is very strange judging by amount of people in this forum who are having issues with this feature. Performance management from RM never worked for me reliably - I do it directly on device.

    Dostałeś, dostałeś - w prywatnej konwersacji, tej samej, której użyliśmy za pierwszym razem. Tym razem się będziemy musieli coś innego niż zoom zorganizować. Macie jakieś propozycje? Skype? Signal? Zoom w bezpłatnej wersji chyba oferuje tylko 40 minut.

    This is very subjective topic and you have a very good chance of getting 50% positive and 50% negative responses. I looked and the info sheet for this pack and is uses just two cabinets. Cabinet is ~80% of tone, so if you don't like sound of Mesa Oversized 4x12 (hard to say which microphones they used - they didn't mention it) - there is a chance you won't like the pack at all. So I won't answer your question directly - this pack might be great, or not..., but I'd recommend considering packs which offer you bigger tonal variety (many different cabs, many mics, different positions etc). For example in high gain territory SinMix's packs cost similar money but you get ~14 cabs and a lot of mics to choose from. There are probably other makers who have similar approach.

    Cześć, trochę sprzętu nagromadziłeś :) - Też jestem bardzo zadowolony z profili SinMixa, który nawet ostatnio pojawił się na naszym wirtualnym spotkaniu. W domu gram wyłącznie na monitorach studyjnych, nie HS8, tylko Adam AX7 i jestem bardzo zadowolony, trzeba tylko przyciąć górę w okolicach 7 - 8kHz - bo inaczej za bardzo wszystko bzyczy).

    I don't. Another audio quality myth is that 24-bit audio will unlock some sort of audiophile nirvana because it’s that much more data-dense, but in terms of perceptual audio, any improvement will be lost on human ears. Capturing more data per sample does have benefits for dynamic range, but the benefits are pretty much exclusively in the domain of recording not human listening.

    I agree: 16-bit resolution gives 65,536 values to represent samples. 24-bit resolution gives 16,777,216 possible values. This is huge improvement and since most (probably all) plugins are using floating point arithmetic, which has limited precision - it makes computation more precise if you start doing it from more precise values. You can export final product to 16-bit and you probably would not hear much difference, but recording and processing in 24-bit depth is a must, in my opinion.

    no, pre/post is for looper placement (you usually want post or output, to be able to change sound for solo vs. rhythm). There is Looper Volume knob (on Stage it is page 4 of System Settings). Maybe you have looper set to Pre - in this case when you morph the sound would be changed to louder. Make sure looper is Post and adjust volume.

    But I think mathematically it doesn't make much sense to go beyond 48kHz for most of digital audio path (Nyquist) and as far as I researched the topic (I'm not a pro, just curious hobbyist) high end plugins, when needed to avoid digital artifacts (aliasing) oversample internally by a lot. Storing projects in 96K just takes more space and brings nothing at all to quality of audio, because all frequencies in human range can be perfectly represented in 48kHz (even 44.1kHz for vast majority of people on this planet). So maybe people are doing this "just in case", but I think they would have hard time to prove scientifically that it makes sense to go beyond 48kHz. It just uses more CPU and disk when less would do. It is mathematics - hard to argue with.

    Just like you can send many analog signals over a single wire (look at cable TV in your house - it sends hundreds of channels simultaneously over the very same coax cable at the same time + internet) you can do exactly the same in digital domain and to be honest SPDIF is the crappiest protocol of them all - just 2 channels using short, quite expensive cable! For example Dante protocol (audio over ethernet) can transfer 1024 channels on a single CAT-5 cable, long distance. But I guess it is off-topic already by a few miles :-)

    RCA cables on have a single center conductor with a grounding sheath around it. I suppose you can get a stereo signal by the center conductor carrying the Left and the grounding shield carries the Right channel but you loose any shielding.

    SPDIF is digital protocol. Kemper is banging bits out to the wire. There is no left and right channel as in case of analog cables. Both channels are traveling on the same wire in a "frame" (frame is some defined structure of digital words) along with some other information, like clock. And it doesn't need to be left/right - can be wet/dry - Kemper is flexible here.