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CJ Jacobson

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Posts posted by CJ Jacobson

  1. 8 hours ago, Bill Ruys said:

    Both sound fine  but I am wondering why I need to record at 48K when using  ASIO and why I would use ASIO anyway.

    You do not need to record at any sample rate. You can and should chose any sample rate you like, as long as your sound card supports it.

    You use any driver mode that works best for your specific PC and DAW.  All driver modes (ASIO/WDM)are written differently, depending on the sound card's manufacturer.

    So you try ALL driver modes supported (except MME, that just for on board sound chips) and see what works best for you and your PC. What works best for me, will no be best for you.

  2. 10 hours ago, Craig Anderton said:

    I think you mean time stretch, not tempo change? This was changed quite a while ago to be in conformance with the Melodyne keyboard shortcut. Hold down Ctrl+Shift while you click and drag.  Or choose F8 for the timing tool. Note that this doesn't work on acidized or REX files, only conventional WAV files.

    Yes, that's what i mean. It does change the tempo of the clip when you stretch it because it can play back slower or faster depending on the direction you stretch it.  At least it did in the older versions of Sonar.

    Thanks,

  3. 3 minutes ago, James Argo said:

    Sorry I don't get your question... Can you explain a bit what you mean... All I know, Ctrl + drag clip(s) will duplicate the clip(s) to desired location in Track View.. but it has nothing to do with Tempo change.

    Sure it does or sure t did. I just did it in Sonar Platinum

    In the Sonar versions, you can press CTL - and move your mouse to the right bottom corner of the clip while pressing the left mouse button and drag it and you can increase or decrease the clips tempo. I do not see that option anymore

  4. 11 minutes ago, Murray Webster said:

    CbB or do I need an old version with VVocal first?

    If i remember correctly, V-vocal was not a VST or DX, so you cannot do a custom install of an older version of Sonar and install it in a newer version. It was integrated into the old version.

    FYI: Melodyne that is included is far far far better than Rolands V-vocal

    But to get all the other plugins (VST and Dx), all you need is the version of sonar you plan to use installed first and then do custom install for the older ones that include the plugins you want

    • Like 1
  5. As your input device, you are referring to an audio interface? Yes? If yes, what are your needs and budget? how many inputs do you need? how many mic pre's do you need? how many outputs do you need/ do you need 5pin MIDI? Do you want USB or Thunderbolt? SPDIF connections? ADAT connections?

    If you are referring to your Studio Monitors, do you want a sub with them? do you prefer 8" or 6.5" speakers/ whats the size of your room?

    CJ

  6. 31 minutes ago, gill said:

    I need to be able to unlink my audio input and output drivers.  I'm using an AI to record vocals, but I want the sound go go through my computer.  My driver mode is in ASIO, but I can't change my input driver to ASIO4ALL, which is what works for me when I use other DAWs.  Help!

    Make 2 different user profiles in your PC. that may work.  One for one sound card and one for the other sound card...

  7. On 4/10/2019 at 10:54 PM, John Nelson said:

    Most dynamic processors have a gain reduction meter.  To visualize by means of a waveform... is highly unorthodox, not something mixers typically do.

    And in the end the advice is always to use your ears and not your eyes.

    +1 , Fabfilter Pro C and L and their EQ's all have before and after waveform views. for me, i tend to just listen to it.

    For me, I do not care what it looks like. If it sounds good, its good to go! It can look like anything and it wouldn't bother me

  8. 20 minutes ago, Twisted Fingers said:

    What can I do to stop the popping and clicking during playback? I'm using the Izotope Production Suite for mixing and I understand that this suite is CPU intensive but my CPU never breaks a sweat.

    What kind of processing are you doing? I know with a lot of their RX plugins, there are different real time playback modes. make sure your not at the highest quality, if you are using those types of plugins. I usually select the first or 2nd playback algorithm, but when i export, i select the highest one

  9. On 4/5/2019 at 5:44 PM, tunesmithers said:

    am using Cakewalk (by Bandlab) and have SD 3.1.2.  I have noticed that when an instance of SD3 is in my Cakewalk file, the Cakewalk file takes a really long time to open.  Once the project is open in CW, and I open SD3 within, I look at the memory gauge (upper right that shows the kit as it’s loading) and it seems to stall.  For instance, I am loading the NY Avatar kit,the memory is showing 580 as the total RAM, but loaded is only showing 120…  then a minute later it creeps to 150, then a little bit later 168.

    Depending on your PC processing speed and RAM IIRC, it can take a minute to load all the samples up.  I remember there being a settings in superior drums for loading samples into ram.

    The faster it loads depends on your Ram specs in your PC. Also, if thew samples are not on your audio drive and on an external hard drive, this can take longer

  10. 18 minutes ago, Mad Musicologist said:

    Only one question @ CJ Jacobson: "You could also have the dropoutsec in your .ini file set to low" - you mean; too low, or should I change it to "low"? Anyway, I did not yet dare to change anything in the .ini file. 

    I meant that it could be set to low, but its fixed, so do not worry about it

     

    20 minutes ago, Mad Musicologist said:

    Is was the LB-EQ.

    Do you mean LP-EQ? The only LB EQ i know is the Neve hardware EQ. Do you have the vst version of that?

    Anyway, if that'LB EQ is glitchy, try out the Fab Filter Pro Q3 EQ. I can run over 100 of them in Cakewalk with no problem what so ever and its a great EQ. you can do mid and side and independent left and right channel edits

  11. 2 hours ago, girldairy said:

    I used to record at home and give it to a mixer. Now I want to export the music(one acoustic track) by myself and found its volume is very low.

    I spent half of the day to try all kinds of solutions (Sonar, Audacity......)but doesn't work, the song just cracked if I turn up the volume.  Any advice?

    It cracks up because you  are clipping it. You cannot exceed above 0dB in thr digital realm. You really shouldn't to go above -0.03dB to-0.06db to leave room for errors in conversions.

    For mastering, you basically just need an EQ --> Compressor --- EQ (sometiems) --- Limiter, those are all you need for mastering. so just learn them inside out and train your ears to your room and wha lah!!!

  12. On 4/6/2019 at 6:52 AM, Mad Musicologist said:

    I have a track almost finished, many instruments, an original audio track with the singer, quite some VSt-FX, at the bottom of the mix I have been "collecting" the audio outs to busses, the last before "Master" is "MasterPreview", where I inserted a VSt-FX for room / space, and the LB-EQ to have the mastering section completely in the project.

    I see a lot for improvement, but lets start here. Gain staging and signal flow.  The common way to do this:

    Tracks go to the Master Bus and Master Bus goes to your Main Outs 1/2.  Sends go to buses and all buses go to the Master bus. 

    What effects are you using for those +25dB spikes?

    Ok, got that out of the way. 2nd, Drop outs are due to improper driver settings and lack PC performance. so what is your sound card, what settings are they set up to and what are your PC specs. If you have a good PC ,it can be as easy as raising your ASIO buffer to from 64 to 128 or 128 to 256 or 256 to 512 or 512 to 1024.

    You could also have the dropoutsec in your .ini file set to low

  13. If you want to use it a a controller to control Fader and bus and send levels, you should get a motorized one. One made for as a controller.  Otherwise, it doesn't remember any of the Fader, Bus, Send and any other settings the project was previously set to and you'll be starting over every time you open and close a project.

    Kinda counter-intuitive, yes?

    • Confused 1
  14. 22 hours ago, garylue said:

    I've been having an issue lately that I am unable to Split Clips.  The session starts off fine with no problems, then at some point I am unable to Split Clips.  If I restart the program, all returns to normal for a while and then it stops working again.  Not sure what triggers it.  Anyone else having this issue, or have any recommendations?  

    Thanks!

    I have not noticed a split clip problem.  I key-binded my 'A' to split clips, as my 'S' is key-binded for Save. But its all the same. I think you have a plugin, driver or setting that is freezing your DAW, and its not the split option.

    So the next question is, what is in this project and specify audio interface  settings, plugins and anything else pertinent  

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