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Discussion in 'Mixing & Song Critique' started by DonnyThompson, Jul 31, 2014.

  • AT5047

    The New AT5047 Premier Studio Microphone Purity Transformed

  1. DonnyThompson

    DonnyThompson Distinguished Member

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    I think that this is a solid initial approach, at least during the "shakedown cruise" you're talking about.

    (FWIW, I'm 4 months into using Pro X Suite, and I'm still "shaking it down".) LOL

    I always prefer to use plugs/processors that have been coded for a particular platform, as opposed to third-party. I'm not saying that there aren't some great third-party plugs out there, I'm just suggesting that, at least initially, until you get more familiar with the new platform, limiting your processing to plugs that are specifically designed and coded for that program is not a bad idea.
     
  2. rjuly

    rjuly Active Member

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    What advantages do you see from using these native-code plugins versus the rest? Is there a perceptible sonic difference or usability difference, or is it more of a matter of keeping your mental desktop tidy and free from distraction? This last thing is as valuable (if not more) for me as pristine sound, since I suffer from quasi-A.D.D like mindset. Very difficult for me not to get sidetracked by all the possibilities. Limits can be very liberating.
     
  3. DonnyThompson

    DonnyThompson Distinguished Member

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    I can't back it up with any scientific quantifiable data. My personal experience, when using some plugs other than the ones developed for a certain platform - has presented "sonic smearing" and some noticeable latency while using some third party plugs, and when switching to similar plugs that were instead stock for that platform, the smearing and latency went away.

    I'm not saying that all third party plugs absolutely can or will do this. I'm saying that I've encountered it in the past. Although, in the interest of honest disclosure, all these moments occurred while working in a 32 bit environment. Whether that had anything to do with it or not, I can't say.

    I'm afraid that's as "technical" as I can get for you. Although - if you are hearing it, I suppose that's really what counts most, and numbers aren't going to matter, unless you are searching for the degree of change.
     
  4. rjuly

    rjuly Active Member

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    Knowing something and being able to prove it are two different things. I can't say I have found this myself, but the only plugins I have used extensively, other than the native-coded ones, are UAD-2, and they tend to impart 'character' as part of their value proposition. it will interesting to compare them to the Magix plugs.
     
  5. audiokid

    audiokid Chris Staff

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    Hey, I need a clarification on this again.
    I dither all the time, but because I am mixing into my master at the destination SR rate and finishing it all up in one pass, maybe that is yet another thing I need not do regardless of me not even hearing a difference. Up to now, I set my dither to the one recommended "POWr1" and haven't gone back in years to evaluate much more on it. I have listened to the different dither options in Samplitude and to be honest, I can't tell a difference. What's with that? Looks like I need to be refreshed here?

    dithering-sequoia-123010-1.png
     
  6. audiokid

    audiokid Chris Staff

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    I like this one too!
     
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  7. rjuly

    rjuly Active Member

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    So, let me understand this... you are dithering in two places in your chain presumably...?

    PC1 dither to D/A(what sample rate & bit depth?) -> analog realm -> PC 2 A/D (what sample rate and bit depth inbound?) -> PC2 mastering processing -> PC2 dither to D/A(what sample rate & bit depth - 44.1 / 16 bit?) -> CD burner?
     
  8. Boswell

    Boswell Moderator Distinguished Member

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    Dither is only needed where you are digitally reducing wordlength, for example, making a 16-bit CD image from a 24-bit stereo master track. It should not be used at other stages in the process, including going from 32-bit floating point to 24-bit integer, as this is (roughly speaking) a change of number format rather than wordlength.
     
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  9. DonnyThompson

    DonnyThompson Distinguished Member

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    Yup. The only time I use dither is on the every rare occasion where I'm doing the mastering for a client who can't afford a true M.E and wants it sent to a CD manufacturing place.

    I will then use Powr-1 on the final output - from 24 bit down to the necessary 16 bit - thus creating the CD image master from which copies will be made.
     
  10. audiokid

    audiokid Chris Staff

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    It appears my setting is set to switch off dither. I can't even hear a difference on or off. I wonder why?
     
  11. Boswell

    Boswell Moderator Distinguished Member

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    It's not easy to hear, particularly if your main tracks are rock/pop/other loud genre. The sort of occasion where it does make a difference is a ppp section in a classical recording.
     
  12. DonnyThompson

    DonnyThompson Distinguished Member

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    It's the accumulative thing as well... if a user is constantly bouncing down, or comping tracks, rendering multiple times, and each time the default is set to dither, you can start to hear a difference.

    Perhaps the reason why you don't hear it Chris, is because you never were a guy who did a whole lot of the above.

    I'm not sure how Samp/Sequoia has their default(s) set for dithering. I think it's "off", but I'd have to look. I'm fairly sure that Sonar and PT have a default setting of no dither... but I don't remember as it's been awhile since I worked on those platforms.

    edit/afterthought - I'd be interested to know - maybe Bos can answer this - is it even necessary to use dither at all - if using the 2- DAW method, where the capture DAW is set for the final SR/BR of the file... so, if the 2nd DAW is set to record at 44.1k/16, is there even a reason to worry about dither?
     
  13. audiokid

    audiokid Chris Staff

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    My setting is set to dither if I change SR via bounce or Export or burn to CD. Otherwise its off. I think you nailed that on a few points Donny, one being my capture is always 44.1. so there is no need for it on.
     
  14. rjuly

    rjuly Active Member

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    Sequoia arrived several days ago - I'm very impressed with it so far, especially the sonics. Learning curve is deep however, and I can see it will require some serious woodshed time.

    Let me start with a question about basic signal routing. I sometimes like to mix to subgroups and record the subgroups as stems. I have bused some tracks to a subgroup, and while I can see them, as soon as I try to record to that subgroup track, the audio signal disappears. Can I bus audio to a subgroup track and record it?
     
  15. audiokid

    audiokid Chris Staff

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    (NOTE: I'm not in my multitrack studio now to test this)

    I'm not quite following you here most likely because I've not tried to record into a subgroup. Just a guess, Subgroups bypass the AD or midi input as it can contain multiple tracks bused into it. Just curious, when trying to record into a bus, is the input AD the same as the pre assigned to it?

    I always create subgroups or AUX as you describe ( Drums, Bass, Guitars, Keys, Vox, Harmonies, efffect) , assign their DA Stems (1/2, 3/4, 5/6, 7/8 etc) to the analog console/summing box or simply, the master bus...
    The bus audio resides on the track lanes. You won't see audio on the subgroup lanes but you can include automation, plug-ins, extended processing there too.

    Did that help?
     
  16. audiokid

    audiokid Chris Staff

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    Woodshed, I like that (y)
    Welcome to the King of DAW's.
     
  17. audiokid

    audiokid Chris Staff

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    NOTE, for the most part, all the online tutorial for both Samplitude and Sequoia are awesome. The "Help" feature is really good too so be sure to look there when you are stuck during those times no one is around for advise.
    Be aware though, some of the "online" tutorials can be confusing for newcomers because they are often not the same versions. You are on Sequoia 13 now , innocently unaware , the layout or GUI can be a bit slightly different. You can find yourself looking for the tab that was moved to another location on the layout.


    To Magix users if you are listening: It would be very beneficial for the growth of this DAW to discuss feature etc on RECORDING ORG. This DAW lacks public exposure big time. So, I encourage us to ask all you can here. But, they do have a good private forum here so until that ever happens,
    http://support2.magix.net/boards/samplitude/index.php?act=idx
     
  18. DonnyThompson

    DonnyThompson Distinguished Member

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    If you are using Sam 11, 12 or Pro X, and following Kraznet's awesome instructional vids on YouTube, he explains how to change your menu layout to make it easier to follow his instructions, which are based on 11's menu layout. It does not effect your version in any way, other than the way that certain tasks are viewed, and in their locations. Pro X changed the menu layout pretty dramatically, so switching to the menu for 11 really helps in following along on Kraz's vids where he explains virtually every facet of the platform in clear, concise, in-depth, yet very easy to follow instructions.

    (Am-munition has 3 episodes alone LOL ;) )

    (Pro X 2 is too new, Kraz hasn't had a chance to do any vids on it yet but he is planning on it).

    The video to revert menu layouts is located here:


    View: https://www.youtube.com/watch?v=jNJy9If656I


    I'm not upgrading to Pro X 2 yet ... and that's advice from several Sam "power users" who's advice I trust; because right now I'm in the middle of three important projects, and I can't afford to face any potential glitches that might occur in the new version that could potentially alter those projects in any way. I do plan on upgrading, but not until I get these projects in the can.

    FWIW

    -d.
     
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  19. rjuly

    rjuly Active Member

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    I am hoping to be able to record at several points along the signal flow paths at the same time. In the DAWS I have used, particularly Reaper and PT, If I send a signal to a channel, via aux or bus, I can click record and have the destination track record a sound file from the audio being routed to it. In this case with Sequoia, what I am trying to do is to create a hierarchy of sub-mixes (stems) that combine down to a stereo channel. This stereo 'master' channel would than be sent to another channel, on which would go all the mastering plugins. That mastering channel would then be assigned to another channel to record the 'Mastered' version. Ideally, as the at each stage of sub-mixing I would like to be able to record the outputs of the subgroups (Drums, Guitars, Bass, Vocals, etc..) and have the audio routed on to the next sub-mix. So... as an example, a source track - say a Kick drum (along with all the other drums) - is routed to a 'Drums' subgroup track(recorded), then the output of that channell is sent to a stereo mix subgroup track(recorded), then likewise to the mastering chain track, then to the mastered track(recorded). This way I should be able to simultaneously have stems for the mix, the mix itself, and a mastered version recorded on the same pass. Of course all of this depends on being able to record the output of a subgroup (input of a subgroup channel).
     
  20. rjuly

    rjuly Active Member

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    ... sorry... I didn't address the first part of your response...

    The process I am describing is internal to the DAW. the signal does not traverse any D/A or A/D. It is quite common among various DAWs to be able to send signals to a channel and record them internally. I just haven't figured out how to do so with Sequoia - likely to be pretty obvious once I know how :)
     
  • AT5047

    The New AT5047 Premier Studio Microphone Purity Transformed

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