des99

From what you describe, I don’t think you did what I suggested…

Bit depth and bit rate are also two different things…

There's some info about how they got that sound here: https://www.soundonsound.com/techniques/classic-tracks-dire-straits-money-nothing Even they couldn't reproduce it again afterwards, so you'll just have to experiment with the tools you have to get something you like...

Make sure your keyboard is transmitting on MIDI channel 1, and that will likely fix your issue.

BTW If someone does still have their Atari ST and a LOG3 and would like to help me with a little favour to try something, please PM. I have an old song file that seems damaged and crashes all Logic versions I've tried to load/convert it with, and has resisted attempts to fix it. It's not really an important song as such, I'm just interested in whether it will load in LAt 2.0.1 or 2.5. As I recall, I ended up running a pre-release beta version of 2.5 for years once Logic Atari was killed (I never got a final 2.5 release version if there even was one - not sure*) which still had quite a few crash bugs that I had to work around. In think I always had to start a new project and then load my actual project as a second project in order to load songs without it crashing. I still have that version to try with too - I just no longer have my LOG3 (or Atari, for that matter). * Edit: So, the status of Lat 2.5 according to a Mark Wherry review from 1997 is that the it *was* released as an upgrade more as a final wrap up for Atari owners, than a brand new version for sale. I don't recall Sound Technology offering me the upgrade though, despite being on their C-Lab/emagic user list for years (even emagic didn't send me the release version, I think they'd basically finished with the ST by that point, and were busing tackling Windows support). So I stayed on that beta version until moving to Logic Platinum Windows 4.6...

It sounds like you made the audio at 48KHz, but are now playing it back in a project set to 44.1KHz. Try setting your project sample rate back up to 48KHz...

A plugin version of the FS1R (and SY99) would be even nicer. I wish Yamaha would start to get more into the plugin market...

Here you go. LP7 format, will load into all versions of LP10. FS1R_lp7.zip

When you have at least one audio channel strip in your project set to a Surround output, Logic changes the output setup to a surround config accordingly.

- Create a software instrument track, and select it. - Open the patch browser. - Type "liver" in the search field. - Select "Liverpool Bass" to add that patch to the instrument track. It's a sampled instrument patch that uses the Sampler, and in the Patch has a bass amp and compressor plugin on it. Remember, searching in the patch browser is context specific - don't try to find a sampled instrument on an audio track, for example. Alternatively, you can just load Sampler in a software instrument track, focus the patch browser on the Sampler plugin, and again search for "liver", which will just load the single sampled instrument into that Sampler plugin, without the additional plugins.

Capacity is up to you and your budget really. Speeds will likely be limited by however you connect it - I'm guessing your Mac probably supports standard USB3 so you won't get the full speed of a modern SSD over that anyway. It'll still be faster than what you have now. I'm a fan of Samsung T7's, personally. That sort of depends on how you save and organise your projects. If your projects are self-contained packages with the audio assets stored inside them, then it's as simple as copying the project file to the SSD and opening it from there. If however you have projects referencing external audio files saved all over your system drive, that might take a bit more work to clean up. But in general, just copy your projects to the SSD, and work on them from there - that's it. if you get an external drive, make sure you format it as APFS first before using it - don't leave it on the Windows filesystem it will likely come formatted as, or more bad things might happen to your files...

Yes - the hardware reports the I/O it has to your Mac (and thus, Logic). There is no additional configuration you need to do in Logic, Logic can output to the outputs your Focusrite tells you Mac it has, or to record from the inputs it tells your Mac it has. Logic, and the hardware is working as expected. However, almost all audio interfaces have an additional layer of complexity, in that they essentially all have internal routing and mixing functions that sit between the Mac, and the physical I/O sockets. While this does make things more complex, it makes the interface more flexible. For example, you can directly route the inputs of the interface to it's outputs, thus monitoring with zero latency or mixing in external hardware, rather than going to and from the computer. Or you can configure the outputs to send to multiple output copies, so for example you send to Outputs 1 & 2 from Logic, but in audio interface, it takes those software returns and outputs them to both hardware Outputs 1 & 2 and also Outputs 7 & 8 to a different monitoring system - lots of things you can do with this kind of flexibility. This is a function of the interface, and is controlled and handled by it's control panel software - Focusrite Control in this case. While by default it will come shipped in a sensible configuration, because you can route and mix and change that setup, you'll need to check things are routed as you expect. Essentially, you mostly want the inputs *not* routed or mixed to the interface's outputs (you usually want them to go to the DAW for recording and software monitoring, generally), and that the software returns (ie, the outputs from the Mac) and going to the same respective hardware outputs (because they don't *have* to be routed like that - there's a layer of abstraction between the software returns - the outputs from the software - and the actually hardware outputs). You can also configure what is sent to each headphone output, as these can have different mixes of source signals too. If you get stuck with the routing in Focusrite Control, likely uninstalling, rebooting, and reinstalling will probably reset your audio interface back to it's factory shipped configuration, which will reset any additional user set routings, and revert to a more sensible default configuration. As said above, you can use the metering on the control panel software to check your incoming and outgoing signals and make sure they are being routed where you expect.

Fusion drives are notorious for performing terribly for these applications, lots of people have problems with them. If you are stuck with it, I recommended getting an external SSD drive and recording your projects to that, and avoid doing any performance-critical work on a Fusion drive if you can.

GForce M-Tron Pro IV Presets from the Mellotron plugin. M-Tron Pro IV.zip Instructions: Once unpacked, Copy the "Factory" folder to ~/Library/Audio/Presets/GForce/M-Tron Pro IV/ (that's the Library folder in your home directory) Index to this thread by Type / Manufacturer / Plugin: Third-party Patches Index

Indeed - digital playback doesn't just playback the raw samples - that's not how digital audio works. The original waveform audio is reconstructed *perfectly* (up to the Nyquist limit) via reconstruction filters. If your audio, for example, has no frequency content above 20K, it doesn't matter whether you play it back at 48KHz, or 192KHz - the reconstructed waveform will be *identical* in both cases (assuming the devices are functioning correctly). There is no "the 48KHz will sound bad" thing going on, due to some kind of "lower resolution" type thing - digital audio doesn't work like that (although there are a lot of myths and misconceptions resulting from an incomplete understanding of what is admittedly not that intuitive if you don't already know how it works.)