Page 2 of 2 FirstFirst 12
Results 11 to 19 of 19

Thread: Sample Rates (44.1 v 48 v 96 )etc etc

  1. #11
    Join Date
    Jun 2015
    Posts
    671
    Thanks
    128
    Thanked 130 Times in 118 Posts
    Sign in to disable this ad
    Quote Originally Posted by chris carter View Post
    But.... and a big one.... But, IMD can shoot through the roof on 96k projects when mixing. So quite often 96k sounds noticeably worse than 44.1k. A lot of problems start creeping in with plugins when you run your project at 96k. You also have to wonder if it makes any sense since virtually all plugins now upsample to 96k or 192k or higher automatically. A few things to consider:

    1) in a proper band limited system there is no conversion difference between 44.1k and 96k
    2) With linear math, the sample rate doesn't matter
    3) virtually all plugins are running at 96k or higher internally
    4) lots nonlinear plugins don't properly filter the incoming 96k signal and will perform better with a 44.1k signal and simply upsampling.

    In other words, the old-fasioned "more is better" doesn't apply to sample rate. It's far more complicated than that.
    Yes it is very complicated and to some degree you are right, but one thing that needs your attention here is the impact of latency and its impact on the typical ITB setup. You need to know precisely what DAW software + version to combine with what audio interface and it is the highest sample rate that in general also gives the lowest latency. Part of the complexity is in that combo. If you use plugins you can account for a certain amount of inaccuracy in the figures fed to the internal delay compensation, so the only way of getting truly low latency is to both limit or minimize it at whatever that has a potential impact on it. One way of doing this is to not use any digital processing at all. If you on top of that use the right software and hardware for the recording, then it is mostly the clock accuracy + sample rate combination that is going to be a concern. Now because of the limits in the amount of available headroom you have in various audio interfaces, it becomes this tradeoff you need to solve mathematically. But from blind tests I have done I know the clock accuracy is of high importance. Various DAW software and plugins on certain platforms can also have long term latency performance drops. I know for instance there are long term latency issues in Pro Tools on the PC platform.

    There are certain signal processing routes you can take to manage it with a certain amount of latency tolerance, but in general if you are really serious you skip the digital processing altogether. In this way eventhough there is a certain amount of long term latency increase natively in your particular setup as you play your mix over and over, it stays within a manageable range.

    My overall recommendation is to put music first, become very hardware aware and use that awareness to improve the quality. This is to some degree also a matter of finding your true signature, that type of vibe that you personally love. To have that integrity I think is key. When using software I think it is best to use an integrated approach in which you distribute the processing.

    You should love what you hear. It should feel very "right" to you...
    Last edited by DarkRed; 09-16-2017 at 02:08 AM.

  2. The Following User Says Thank You to DarkRed For This Useful Post:


  3. #12
    Join Date
    Jul 2010
    Location
    Australia
    Posts
    408
    Thanks
    73
    Thanked 52 Times in 52 Posts
    Quote Originally Posted by DarkRed View Post
    Yes it is very complicated and to some degree you are right, but one thing that needs your attention here is the impact of latency and its impact on the typical ITB setup. You need to know precisely what DAW software + version to combine with what audio interface and it is the highest sample rate that in general also gives the lowest latency. Part of the complexity is in that combo. If you use plugins you can account for a certain amount of inaccuracy in the figures fed to the internal delay compensation, so the only way of getting truly low latency is to both limit or minimize it at whatever that has a potential impact on it. One way of doing this is to not use any digital processing at all. If you on top of that use the right software and hardware for the recording, then it is mostly the clock accuracy + sample rate combination that is going to be a concern. Now because of the limits in the amount of available headroom you have in various audio interfaces, it becomes this tradeoff you need to solve mathematically. But from blind tests I have done I know the clock accuracy is of high importance. Various DAW software and plugins on certain platforms can also have long term latency performance drops. I know for instance there are long term latency issues in Pro Tools on the PC platform.

    There are certain signal processing routes you can take to manage it with a certain amount of latency tolerance, but in general if you are really serious you skip the digital processing altogether. In this way eventhough there is a certain amount of long term latency increase natively in your particular setup as you play your mix over and over, it stays within a manageable range.

    My overall recommendation is to put music first, become very hardware aware and use that awareness to improve the quality. This is to some degree also a matter of finding your true signature, that type of vibe that you personally love. To have that integrity I think is key. When using software I think it is best to use an integrated approach in which you distribute the processing.

    You should love what you hear. It should feel very "right" to you...
    I have a Steinberk UR MK II, which they say has zero latency for vst.. as I think that is it's design.. it does a pretty good job...

    I Master in Pro Tools.. and signal everything to busses, and then the main mix bus.. I use SSL compression on the parallel compression bus, and send that back to the mix bus, I send it to two mono tracks and split them to stereo to widen the image with no effects, I send that back to the mix bus..

    On the mix bus, depending on what it needs, but generally I have mixed it, and a lot of the work has been done there, so i will glue it with linear phase EQ, as if linear phase is all on the main mix bus, it send the whole signal out linear to the next thing, like tape saturation, and multi band limiting.. not much, just touching the threshold... then a limiter oversampled with a graphic display, so see what is happening there, then another limiter at 0, just to stop any possible peaks after that.. I have achieved the best sound yet on my latest track... it's close to professional reference tracks now..

    I learn pretty quick these days, but the learning curve is steep.. and a long journey to it's peak so to speak ;P
    Nobody wins this game......

  4. The Following User Says Thank You to Synapsis For This Useful Post:


  5. #13
    Join Date
    Jul 2010
    Location
    Australia
    Posts
    408
    Thanks
    73
    Thanked 52 Times in 52 Posts
    Oh, and another little treasure, if you are limited to mixing/mastering in headphones, a handy plug in called redline monitoring creates stereo speakers and room ambience in your headphones, and does a fantastic job, you can check the phase and everything
    Nobody wins this game......

  6. #14
    Join Date
    Aug 2009
    Location
    Los Angeles, CA
    Posts
    1,265
    Thanks
    0
    Thanked 59 Times in 41 Posts
    Sometimes I don't know why I bother.....
    Chris 'Von Pimpenstein' Carter - Hit Producer & Mixer with three #1 hit singles
    http://www.vonpimpenstein.com

  7. The Following User Says Thank You to chris carter For This Useful Post:


  8. #15
    Join Date
    Jul 2010
    Location
    Australia
    Posts
    408
    Thanks
    73
    Thanked 52 Times in 52 Posts
    Quote Originally Posted by chris carter View Post
    Sometimes I don't know why I bother.....
    https://www.kvraudio.com/forum/viewtopic.php?p=6462080

    Neither do I..... It's a headfuck all that shit
    Nobody wins this game......

  9. The Following User Says Thank You to Synapsis For This Useful Post:


  10. #16
    Join Date
    Jul 2010
    Location
    Australia
    Posts
    408
    Thanks
    73
    Thanked 52 Times in 52 Posts
    480 is the most common, but by no means a standard.

    Here's what I've found so far:

    Logic - 480
    DP - 480
    Cubase - defaults to 480, but adjustable
    Sibelius - defaults to 256, but adjustable from 96 to 960
    Finale - 1024
    Ableton Live - 96

    Note that these are for exported MIDI files only, almost all of these programs run at a different internal resolution. I'm not sure if all of these numbers are 100% accurate, some of them were found on forum posts, not in manuals.

    I'm really surprised at how difficult it is to find this information. I have a way to figure this out, but thought that searching the internet would be faster. Guess I was wrong. If no one confirms this I will post my findings here after I run some tests.

    I think that PPQ, has something to do with sample rate somehow... I mean, it's 96 ppq, 960, 480.. I wonder.. I think I will stay above 44.1, just for my own sake..



    Nobody wins this game......

  11. #17
    Join Date
    Jul 2016
    Location
    Washington State
    Posts
    25
    Thanks
    3
    Thanked 4 Times in 3 Posts
    I've been recording, mixing, mastering at 44.1/24 for several years and would like to move to 96/24, but I'm afraid my laptop won't handle it well, especially wrt latency. My processor is i7-4500U @ 1.8GHz, with 16GB RAM, which handles up to 50 tracks fine in my DAW with no noticeable latency at 44.1/24. Does anyone know what I'd need to support a similar load at 96/24? I've seen some articles which say you need Quad Core 3.0 or better, but I'd like to hear from someone who has actual experience...

  12. #18
    Join Date
    Jun 2015
    Posts
    671
    Thanks
    128
    Thanked 130 Times in 118 Posts
    Quote Originally Posted by JmgWeb View Post
    I've been recording, mixing, mastering at 44.1/24 for several years and would like to move to 96/24, but I'm afraid my laptop won't handle it well, especially wrt latency. My processor is i7-4500U @ 1.8GHz, with 16GB RAM, which handles up to 50 tracks fine in my DAW with no noticeable latency at 44.1/24. Does anyone know what I'd need to support a similar load at 96/24? I've seen some articles which say you need Quad Core 3.0 or better, but I'd like to hear from someone who has actual experience...
    For anyone who is serious you need a dual cpu solution, with 4 cores on each processor at a minimum. Then on top of that, you want as much as possible of the plugins to do their processing on their own dedicated hardware. Then on top of that you also ensure you have as much RAM and SSD drive capacity as possible, especially RAM. I think Mac Pro computers are great for DAW use.

  13. #19
    Join Date
    Aug 2008
    Location
    France
    Posts
    1,092
    Thanks
    5
    Thanked 32 Times in 31 Posts
    It's only good for softwares managing the multi-processing correctly.
    Mastering 2 per minute - 7/24/365 - 1 Hour turnaround - Free test - Try now
    MaximalSound.com

Thread Information

Users Browsing this Thread

There are currently 1 users browsing this thread. (0 members and 1 guests)

Posting Permissions

  • You may not post new threads
  • You may not post replies
  • You may not post attachments
  • You may not edit your posts
  •