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Jim Roseberry

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Posts posted by Jim Roseberry

  1. On 7/6/2019 at 9:47 AM, Max Arwood said:

    Like Samplitude. Which is way behind on midi  But has a few really nice Audio features I wish were in sonar. Like the one Jim is talking about. 

    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.

    • Like 1
    • Great Idea 1
  2. On 7/2/2019 at 4:45 PM, razor7music said:

    PS - I was referring to additional tracks (instruments, layers of instruments) versus comping different song parts on different tracks. So, if my arrangement has 4 parts, say verses, pre-hooks, choruses, bridge, then other than the drums, more times than not I'm going to have 4 tracks of each instrument. I do that so if I need to tweak anything on an instrument for a certain part of the song, I can easily do that on that respective track.

    Just clarifying--but I see most of you knew what I meant already.

    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)

     

    • Like 2
  3. 1 hour ago, Blogospherianman said:

    That’s what Process, Apply effect, Gain does,  It will Apply Clip gain to the clip or selection and scale the waveform.  That’s the way I do some pre-leveling of lines, words or syllables.

    FWIW, I'm well aware of the destructive processing option.  😉

    The static gain change to which I'm referring is non-destructive.

     

    • Like 2
    • 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.

  4. 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.

  5. 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.

     

  6. +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.

     

     

  7. 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).

     

  8. 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.

  9. 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.

     

     

  10. 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.

     

     

  11. 12 hours ago, will stahr said:

    OK, I have been using Cakewalk since it was released for free by BandLab on an HP AMD A8-5500 "quad-core" machine  (really like dual core with HT if it were Intel) with like 12GB of DDR3 and a SATA SSD without any issues. I decided one day recently that I would like to mix songs at my work computer where I have a 55" 4K TV to work on. My "work" machine is an Intel i7 8700K, 16GB DDR4, M.2 NVME SSD and an RTX2070 so no slouch. Yet for some reason, cakewalk has like a 3-5 second delay when doing anything on my work computer, click and wait 3-5 seconds, click and wait another 3-5 seconds, over and over. It never speeds back up to normal, really annoying and renders the software unusable. What is interesting is if you get into the settings menus it works like normal without the delay. I have uninstalled and reinstalled Cakewalk, made sure all my drivers were up to date and even went the extent as to reinstall windows from scratch. None of which fixed the issue, I'm an IT system admin by trait so I have done a bit of troubleshooting before asking for help. I should note I also recently upgraded my studio machine to an MSI laptop with a Intel i7 6700HQ, 16GB DDR 4, M.2 NVME SSD, and a GTX1060 which the software runs perfectly on. All three machines are running the latest version of Windows 10 (1903 at the time of writing this.) I'm at a loss and really want to be able to do some mixing at my main workstation, please help!

    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.

    • Like 1
  12. 16 hours ago, Jim Thomas, said:

    This could be addressed part of a more general feature request for hardware profiles:

    If you create an aggregate device (using separate audio interfaces), keep in mind that each is using a separate clock.

    Over time, slight timing differences between the clocks will cause audio tracks to drift apart.

    You want all audio interfaces sharing a common clock (via BNC, S/PDIF, or Lightpipe)... so there will be no drift.

     

    • Like 1
  13. Windows 10 is a fine mature DAW platform. 

    We've got many DAW using clients running it (including professional composers/engineers/musicians).

    You need to be diligent with backups (should be doing that anyway)... and it certainly helps to have the Pro version (Group Policy Editor and Registry tweaks to stop all Automatic-Updates.  Once reined-in, there's no issue at all with Win10 for DAW (or video) purposes.

     

    If you want to run Thunderbolt (using "PCIe via Thunderbolt" for PCIe level performance), you have to be running Win10.

    Microsoft doesn't support "PCIe via Thunderbolt" under Win7.

  14. Win10 is an excellent DAW platform (once reined-in). 

    The Pro version is helpful in this regard... as the Group Policy Editor makes it quick/easy to disable Cortana, OneDrive, etc.

    With the Pro version, you can also add two Registry entries that'll stop all Automatic Updates (including notification).

     

    Win10 boots extremely fast here (not slower than Win7).

    Modern Z390 and X299 motherboards boot fast.  Use an SSD as boot drive... and the machine boots extremely fast.

     

    When applying major updates, check to be sure DAW tweaks are maintained.

    Settings like Fast Startup... and power-management features (that were disabled) can be re-enabled by some updates.

     

    • Like 1
  15. We were going to stay most of Saturday... but the wife and I were exhausted after being there for 12 hours Friday.

    Stayed until about Noon on Saturday... and headed home.

     

    This year was more crowded than the past several.  I'm guessing 15,000 people.

    Paul Alfery (one of my clients) was doing demos for Music Group.   Was great to meet him and talk shop.

     

    Wanted to get to both of Craig Anderton's seminars... but unfortunately missed them both.

    Got to speak with Craig for a few minutes while we were outside perusing the Presonus tent.

    Even tech-savvy folks learn from Craig; always something to add to your work/methods.

     

    Sweetwater puts on a great event.  This year was no exception.

     

    • Like 1
  16. I wasn't referring specifically to the 3950x... (just meant in general that's what I'd like to see from either company)  😉

     

    Unless there's been a radical change, there's no way the 3950x will do 5GHz across all 16 cores.  Thus far, Ryzen has had limited over-clocking ability.

    I doubt the 3950x will do full boost clock speed (4.7GHz) across all 16 cores.  If it can... then we've got some serious competition.

    If the 3950x averages ~4GHz for each core, (to me) that's not overly exciting. 

    I want to see the performance envelope pushed.  🤪

     

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