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Jim Roseberry

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Everything posted by Jim Roseberry

  1. For DAW purposes, clock-speed is absolutely the single most critical factor. Not all processes in a DAW can be multi-threaded. ie: Playing/monitoring in realtime thru an AmpSim plugin at 96k using a 32-sample ASIO buffer size is not something that lends itself to being heavily multi-threaded. Virtual instruments like UVI Falcon only use a single core. In a perfect scenario, you want highest possible clock-speed... AND the highest number of cores you can get. What you absolutely *don't* want to do is choose more cores... at the expense of significant clock-speed. This is why Xeon CPUs (even though they're more expensive) are usually a significant performance hit compared to standard CPUs. They have more cores... but typically significantly lower clock-speed. Right now, this is why the Intel i9-9900k is such a great "sweet-spot" for a DAW. With the proper configuration, you can lock all 8 cores (16 processing threads) at 5GHz. That's super high clock-speed... and 16 virtual cores (8 physical cores). With quality air-cooling, the 9900k will do the above while running near dead-silent. To best the 9900k, you have to go high-end socket-2066 i9 (which is $1400+ just for the CPU).
  2. During his original run with Journey, I thought Steve Smith was good. After leaving Journey, his level of playing went up dramatically. He became one of the best players on the planet. His playing is smooth/fluid... looks and sounds effortless.
  3. A MIDI driver that's not "letting go" can cause CbB (Sonar) to not shut down. A while back, Korg's USB MIDI driver was causing this issue. When you close CbB (Sonar), open Task Manager and see if the Sonar process continues to run. Some plugins like Addictive Drums have an online "Sync" process. This can cause CbB (Sonar) to take slightly longer to shut down.
  4. Thunderbolt works just fine on a PC (if you know how to properly configure it). A MacBook Air is a particularly bad choice for DAW purposes. The reason is simple, its CPU has *slow* clock-speed. When choosing a CPU for DAW purposes, the most critical factor is clock-speed. Not all processes in a DAW can be multi-threaded (spread across cores). Things like playing/monitoring in realtime thru an AmpSim plugin at 96k using a 32-sample ASIO buffer size (super low round-trip latency)... don't lend themselves to being spread across multiple cores. Some plugins don't make use of multiple cores. ie: UVI's Falcon More cores is certainly beneficial... but not at the expense of significant clock-speed. This is why Xeon CPUs (although more expensive) are often a significant performance hit... compared to standard CPUs. They have more cores... but (typically) significantly slower clock-speed.
  5. I'll check that out... 😉
  6. I can't bring myself to use one of those... though the Dr was recommending it for sinus issues. Feel like I'm drowning myself...
  7. The wife absolutely loves hers.
  8. I agree... Samplitude started out as a really advanced (realtime) audio editing application. It had many realtime processing options decades ago... which was revolutionary back then. I'd LOVE to see that "Static Clip Gain" control added to CbB. I'm mixing a project for a client as we speak... and it would certainly help speed up the process. Yes, I can work around it... but it's not the same. It would be a huge time saver.
  9. I think that's the way many are working (spreading sections of each part across tracks (for separate processing of each section, etc). This is why many productions are using more than 24 tracks. If you listen to any one point in the song, there's usually not that many different parts playing simultaneously. With BG Vocals, it's common practice to triple-track each harmony part (left, center, right). This helps a couple of voices sound more like a group of singers. If you've got three-part harmony... and put each section on separate tracks, the track count quickly adds up. In the case of mocking strings/winds/brass, it adds realism to have each part tracked individually. In my case, when doing "punch-ins" to fix a section, I don't like punching-in on the original track. All my punch-ins are recorded on separate tracks. I prefer DAWs that allow processing per-clip in addition to per-track. DAWs that don't allow processing per-clip force the user to spread those parts across multiple tracks. I do think there's also the, "Because we can!" factor. With the processing power available today, folks are going to use it... (for better or worse)
  10. If turning off Input Echo in CbB reduces the "hiss", then the issue isn't Melodyne... but rather the monitor chain.
  11. FWIW, I'm well aware of the destructive processing option. 😉 The static gain change to which I'm referring is non-destructive.
  12. I've been lobbying (for a good while) for a static "Clip Gain" parameter. 😉 This static clip-gain parameter would ideally scale the waveform up/down. This makes it quick/easy to level out the volume of tracks. Clip Gain Envelope works... but it's a slower process (and no waveform scaling).
  13. Not all processes in a DAW can be multi-threaded (spread across multiple cores). Some virtual-instruments (like UVI Falcon) don't make use of multiple cores. Playing/monitoring in realtime thru an AmpSim plugin using a 32-sample ASIO buffer size is not something that can be heavily multi-threaded. For these and other similar reasons, it's not always possible to have completely balanced load between cores. This is why CPU clock-speed is still the single most important factor when it comes to DAW performance. More cores is certainly beneficial... but not at the expense of significant clock-speed.
  14. If you have a library that loads particularly slow, put that library on a M.2 Ultra SSD. M.2 Ultra SSDs use 4 PCIe lanes... and sustain ~3500MB/Sec SATA SSD sustains ~540MB/Sec Conventional HD sustains ~200MB/Sec This will allow projects using the Ravenscroft 275 to load much faster. FWIW, I had the same issue with HALion 6's library (loaded slow). Put the library on a M.2 Ultra SSD... and it loads fast.
  15. So sorry to hear this! Thoughts and prayer for you and your family
  16. Only worth it... if you use it. 😉 Neve channels are (IMO) one of the best gear investments you can make... But I'd only recommend to those who are into recording/mixing for the long-haul.
  17. I've brought this up in numerous posts recently. If you compare playing thru a real guitar amp vs. plugin thru an AmpSim plugin, the real amp responds more smoothly to transients. Craig Anderton has written about this in numerous articles. Craig is the reason why current Les Paul Standard HP models have the dip-switch option to reduce these types of transients. Using a compressor on the way in (to record) can help the AmpSim respond/sound more like a real amp. I agree with CJ in that you've got to get your guitar sound "up front". If you're doing much more than using a high-pass filter and maybe a subtle EQ boost/cut, the sound (up front) isn't right. Same with live guitar tone; If the sound engineer is using radical EQ, he/she is doing more damage than good. If you've got a nice studio channel, that can make a massive difference in the quality of result from an AmpSim plugin. Something like the Neve Shelford channel (world-class DI/preamp, quality EQ, and versatile compressor) is ideal for use with an AmpSim plugin. Of course, this type of channel-strip costs as much as a (real) quality guitar amp/cab. The DI/preamp alone can make a very significant difference in tone. ie: A passive Fender P or J bass recorded thru a cheap DI sounds weak/anemic. That same bass recorded thru a Neve DI/preamp sounds amazing. The tone is just there... no struggle. Used lightly, the compressor will help the AmpSim mimic the way a real amp responds (more smoothly) to transients. Finally, the EQ section can do wonders to shape the guitar tone. ie: Dial up a Friedman BE-100 amp model (popular Marshall clone). Set the Shelford's mid EQ to the 1.8k setting... and add a slight boost. The resultant sound is a great "Mid pushed" Marshall tone. I can't stress enough just how significant the difference from using quality DI/preamp. Using a typical cheap DI makes it much more difficult to achieve great guitar/bass tone.
  18. +1 on the suggestion of using gear with (real) analog transformers while recording. This is part of why I like Neve preamps. The transformer gives the sound some "girth"... but in a different way from a tube (not as "soft/squishy"). You can over-do harmonic distortion from either tubes or transformers... making the mix muddy. Newer Neve models (Portico-II or Shelford Channel) have a knob that lets you dial in the desired level of transformer distortion (they call it "Silk"). The "Red" Silk setting adds harmonic distortion that enhances higher frequencies. The "Blue" Silk setting adds harmonic distortion that enhances lower mid frequencies. As with a sonic "Enhancer" or "Exciter", it's all too easy to over-do the effect. I make use of the Silk function if the track needs some extra "sparkle" on the high end... or needs filled out in the lower mids. For many tracks, I leave the Silk setting off.
  19. That's the ProTools equivalent of your "VST Plugins" folder. You'll be fine without it... as long as you're not wanting to run ProTools.
  20. I'd start at the source: Guitar Preamp AmpSim If you get the sound pretty close "up front", it's a whole lot easier to mix. A little high-pass filter on the bottom end (to keep from competing with the kick/bass)... and maybe a minor EQ boost/cut. If you're EQ'ing the guitar heavily at mix, the signal (up front) isn't right. A quality DI can make a huge difference in the final result using software based AmpSims. Something like the Neve Shelford channel (albeit expensive) can make a massive difference in guitar/bass recordings. I was just talking about this on The Gear Page. The Shelford channel combines a world-class Neve DI/preamp, EQ, and versatile compressor. If you've ever tracked a passive Fender bass thru a cheap DI, it sounds weak/anemic. That same passive Fender P or J bass sounds great straight off the Shelford channel's preamp. If you've ever compared playing a real amp vs. playing thru software AmpSims, the real amp responds much more smoothly to transients. The Shelford channel's compressor can be used to smooth out the transients... making the AmpSim respond/sound more like a real amp. I know Craig Anderton has written about this subject (using a compressor before AmpSim) in numerous articles. He's also the reason why Gibson Les Paul Standard HP models have the dip switch position to reduce these transients. Finally, there's the EQ section... which is perfect for tone-shaping on the way into the AmpSim. Dial up your favorite Marshall tone. Now, engage the Shelford's Mid band and give a slight boost at the 1.8k setting. Perfect for that "pushed Mid" Marshall tone. Running out and getting a world-class channel-strip isn't practical for every situation... but it's one of the few things that can make a very significant difference. As with all things recording, get the sound as close to "right" as possible... up-front (at the source).
  21. Hi Neil, I've been using TRacks 5 plugins for a good while (in particular the 1176, LA2A, Pultec, Tape Delay, and Stealth Limiter). FWIW, I haven't encountered any CPU load issues. When choosing a CPU for DAW purposes, CPU clock-speed is the single most important factor. Having more cores is beneficial... but not at the expense of significant clock-speed. This is why Xeon CPUs (although expensive) aren't a great choice for a DAW.
  22. That's a MIDI performance that you dragged into Cakewalk's timeline. I'd open Addictive Drums and click on the individual "kit" elements (drums/cymbals)... to make sure you're hearing them. If you can't hear them, it's almost surely a signal routing issue. ie: lf you load AD on an audio track (instead of an Instrument track), you have to put the MIDI performance on a separate MIDI track... and route that MIDI track to that particular AD instance. More complicated... but more flexible. An "Instrument track" (specifically for using virtual-instruments) combines an Audio and MIDI track. You assign AD... and drop the MIDI performance on that track.
  23. When using plugins like Falcon (that don't use multiple cores), you may achieve better performance using multiple instances... rather than a single instance playing all parts. This would help spread the total load across cores.
  24. There are numerous potential reasons for a single core being under heavier load than the others. Not all processes in a DAW can be multi-threaded. ie: Playing/monitoring in realtime using a 32-sample ASIO buffer size is not something that lends itself to being heavily multi-threaded. Some plugins don't use multiple cores (ie: UVI Falcon). If you've got a heavy load running in Falcon it's going to result in one core being heavily loaded. The lower the clock-speed on your CPU, the more single-core spikes will be evident.
  25. The issue isn't Cakewalk. Sounds like you're using the onboard sound on the motherboard for your audio interface. You can lower buffering (in Cakewalk) to minimize the latency, but it's not going to be ideal. For responsive playback/recording, you want to use a dedicated audio interface that has a proper ASIO driver. A dedicated audio interface will also have significantly lower noise-floor, better A/D D/A, etc. Also, though the RTX-2070 is a great video card, it has been causing high DPC Latency. High DPC Latency can cause glitches when working at low (audio) latency settings.
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