blinkofani

Pretty sure it’s marketing. It looks better in ad documents to read that Logic Mac 11 is compatible with Logic iPad 2 than Logic Mac 10.8.2, hihi. The 11 thing worries me a bit though as Logic 10.0 was the worst, most bug-ridden software I’ve ever use. But since it’s free to existing users, I’d venture there’s not so much novelties under the hood as what happened between 9 and 10. Interesting stuff though. Can’t wait to read the release notes next week. I’m not much into AI stuff but I’ll see how it behaves before making judgement.

They’ll always only advertise the new shinny stuff on the product page. Even though I’d be much more excited with something like « Logic Pro now puts the color palette right under your cursor when the key command for it is triggered », it doesn’t make up for great marketing stuff!!

Crazy!! available next week https://www.apple.com/newsroom/2024/05/logic-pro-takes-music-making-to-the-next-level-with-new-ai-features/

Well, I could if you insist, as I still have a G5 Mac with Logic 9.0.2/Leopard up and running, plugged to the DVI input on my main monitor. But the G5 have been used as a keyboard stand for a Korg microKORG in the recent years, fits perfectly!! But to use the word « running system» wouldn’t be true! 😀

So, the number of recordings per second(sample rate) has an impact on the quality of the end result depending how many cycle a pitch have(your example of a sine wave cycling at 20K)? But most sounds in life are more complex than a sine wave. Anyway, I’m not trying to argue with you and what you write(it seems to sound like that!!), I just don’t understand the link there can be between the speed of sampling and the pitches that will be retained!! For me, it seems logical that the more recordings every second, the more precise the rendering and that it has no impact on the frequency spectrum captured. I hope I make sense. Being french Canadian, it’s already a bit of a chalenge trying to explain myself in that topic!! Thanks David.

Thanks for your explanation, but I still don’t see how the sampling frequency(the number of recordings every second) has anything to do with this. When we use an EQ(filter) it’s for manipulating the content of high/low frequencies into the sound. When we change the sampling frequency when recording, it’s affects the number of recordings happening every second, nothing to do with the timbre of the recording. Anyway, have to go to bed now!! But i appreciate your time. Maybe it’s me who’s not smart enough to see the link here.

So, you’re saying more samples(recordings) will not render the original signal that much better because of the rounding off of the filter? This i kind of understand, but why are people saying that it’s related to the frequencies the human ear can hear?

So, today is the day i ask the question that’s been haunting me for a long time. I could have ask in any audio-related forum but since the ratio of smart people is rather high here, I thought it would be a good place!! So, we often hear that it’s doesn’t make that much of a difference to record above 44.1KHz because of « I don’t remember the name of » the law you divide by 2, that makes that frequencies above 20KHz won’t be captured or something and most human ears can’t hear above 20K. Right? It’s just useful to dogs to use a higher sampling rate, bla bla bla. But, when we talk about frequencies the human can hear, we’re talking about frequencies related to pitch. When we talk about sampling frequencies we talk about the number of samples(recordings) a converter will make every second. What relation does it have with the highest frequencies a human ear can hear? In another kind of sampling, if I take 4 pictures a second of someone walking in front of me, and then 16 pictures a second, the later will give a smoother idea of the walk but I won’t see less content in the pictures, no? Don’t know if I’m making myself clear? What’s the relation between the frequencies in Hz that the ear can hear(pitch) and the frequencies of a recording(number of samples captured every second)? Thanks for any explanation.

Thanks for the quick reply. That's great, will saves me some time!!

Hi all, for some time now I decided to return to an older system using older software but I find the Mastering assistant in LP 10.8 a great tool. It is installed on my current M1 Mac but since I don't use that computer to make music, it's not available on my older Mac. Being 56, I know my ears are not what they used to be, so if I produce on an earlier version of Logic and would want to import it on LP 10.8 just for the final mastering stage, will Mastering Assistant do a better job if I import only a stereo bounce from the project made elsewhere or should I bounce everything in audio, including insert FXs (since some are not available on my M1 Mac), and import the multitrack session in LP 10.8? Does it make a difference for the analysis if the plugin(can we call it a plugin??) has the individual tracks vs the stereo mix down? If MA works better with multitrack content vs a mix down, it would be worth the time to bounce all tracks before importing in LP10.8. Thanks for any suggestions.

You can of course assign a KC to individual tools. But in the case of the Marquee, it is I think common practice for most Logic users to assign it as your Command tool. So no need to do anything apart pressing Cmd and selecting something. Some people assign it to the lower zone tool but never found much love to this solution.

Pro Tools doesn’t use the efficiency cores on Mx machines. I have 8P and 2E on my M1 Pro and PT only sees 8 cores. And if you saw the same video as I did about different DAWs performance, the problem on that video is the user is using a 1024 buffer size like in the old days. But it’s been pretty much proven that Mx machines perform better on lower buffer sizes. Don’t ask me to explain it but that’s what I’m seeing also on my computer.

That was worth a shot. IIRC, that mode was created for your exact use case(loads of input recording at once).

Did you try what I wrote a couple of days ago? Quoting myself: Are you using the Playback+Live Tracks setting in the Audio Prefs or just the default Playback setting?

Are you using the Playback+Live Tracks setting in the Audio Prefs or just the default Playback setting?