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Cristiano Sadun

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Posts posted by Cristiano Sadun

  1. I've been using CW for ages with a rock solid RME PCI setup. Lately I've moved and I haven't set up yet for the outboard and the 18 channels of I/O that were needed for them before. So for the moment I've set up one pc as all-in-the-box setup with one of my USB interfaces, an old UAD Apollo Twin mk I - which so far has been working just fine for non-production tasks. 

    Today I thought of making a demo and opened Cakewalk. 

    What I experience is that playback works fine, but when the transport stops, the software hangs for several seconds.
    This happens also for a very minimal project with just EZ drummer loaded up in an instrument track and nothing else.

    Some facts:

    • Cakewalk version is 2024.07 (not the most recent update, but fairly recent)
    • The UAD driver/console/plugins version is also recent (11.5.0 10-03-2024 build 173918)
    • I'm using the ASIO driver (but the behavior is the  same with wasapi ex for example).
    • The RME PCI card is still in the pc, but the RME DSP driver is not loaded in Windows (regardless the Twin uses USB)
    • The same PC with Cakewalk was running perfectly with the RME PCI interface.
    • I've assembled this setup in a bit of a hurry so I'm not positive the Twin is connected to a USB3.0 port but it shouldn't really matter as 3.0 only gives more bandwidth and I've just one track.
    • It does not matter if the playback stops because I push the "stop" button or because CW reaches the end of the track - the hang always happens
    • CW does not crash - it simply hangs. Is 

    I've not used this particular PC  with a USB interfaces before (other than as a platform to run UAD plugins), but I've used other pcs with various USB interfaces I own, and never seen this kind of issue.

    Any ideas? Also a couple of specific questions:

    1. Does this kind of "hang for a while after playback"  sound familiar to someone?
    2. Is there any parameter in the configuration file that's worth looking at tweaking? My current AUD.INI looks like this (after a reset)

      [Universal Audio USB (11 in, 5 out)]
      InputLatencyOffset=0
      UseAsioReportedLatency=1
      MME.DriverMap.UseWaveOut1=1
      MME.DriverMap.UseWaveOut2=1
      MME.DriverMap.UseWaveOut3=1
      MME.DriverMap.UseWaveOut4=1
      MME.DriverMap.UseWaveOut5=1
      MME.DriverMap.UseWaveIn1=1
      MME.DriverMap.UseWaveIn2=1
      MME.DriverMap.UseWaveIn3=1
      MME.DriverMap.UseWaveIn4=1
      MME.DriverMap.UseWaveIn5=1
      MME.DriverMap.UseWaveIn6=1
      MME.DriverMap.UseWaveIn7=1
      MME.DriverMap.UseWaveIn8=1
      MME.DriverMap.UseWaveIn9=1
      MME.DriverMap.UseWaveIn10=1
      MME.DriverMap.UseWaveIn11=1
      [Aud]
      DataDir=C:\Cakewalk Projects\Audio Data
      PictureDir=C:\Cakewalk Projects\Picture Cache
      [Wave]
      DefaultSampleRate=44100
      DriverID=0
      WaveInID=0
      OpenInputFirst=0
      SmpteMode=1
      TimingOffsetMsec=0.000000
      TimingOffsetBuffers=0
      LatencyMsec=23.219955
      BounceBufSizeMsec=0
      AlwaysStreamAudioThroughFx=1
      MaxInputChannels=16
      MaxOutputChannels=16
      ThreadSchedulingModel=1
      TransDetectorModel=2
      TransDetectorUpgraded=1
      EnableMixThreads=1
      MixThreadCount=0
      EnableSetThreadIdealProcessor=1
      EnablePluginLoadBalancing=0
      MinPluginLoadBalancingBufferSamples=128
      CSUseSpin=1
      AllowOfflineRenderMixThreads=1
      UseMMCSS=1
      EnableMMCSSforASIO=0
      MMCSSThreadPriority=2
      MMCSSTaskKey=Pro Audio
      FreeMemOnUnload=1
      AlwaysOpenAllDevices=0
      MinimizeDriverStateChanges=1
      RemoveDCOffset=0
      EnableAsioBufferSwitchTimeInfo=1
      EnableDeviceOutputLatencyCompensation=1
      UseHardwareSamplePosition=1
      BitsPerSample=24
      FileBitDepth=16
      RenderBitDepth=32
      ImportBitDepth=0
      ExtraPluginBufs=0
      MixDezipperUsec=50
      GapDezipperUsec=500
      WaveInBuffers=8
      WaveOutBuffersMME=4
      WaveOutBuffersKS=2
      MeterFrameSizeMS=40
      SyncMaxDriftMsec=2
      SyncDivisor=8
      ProfiledMME=0
      ProfiledKS=0
      ProfiledWASAPI=0
      UseWDMDmaForWASAPI=1
      LinkSendPan=0
      LinkPFSendMute=0
      StopOnEmptyPlayQueue=0
      KsUseInputEvent=0
      WaveOutExtraBuffers=1
      AutomationDecimationMsec=5
      EnableSSEMixing=1
      ThumbnailCacheSize=100
      EnableLiveADCRecalc=1
      UseAlias=0
      DropoutMsec=250
      MaskDropoutDetection=0
      StartFadeMsec=0
      StopFadeMsec=0
      PanLaw=0
      PanLawCompatMode=0
      DisableIMDuringPlay=0
      ShowMultiChannelInputs=1
      ShowMultiChannelOutputs=1
      MaxPreviewMsec=300000
      EnableWasapiDSP=1
      RestartEngineAfterDropout=1
      [SampleRates]
      Count=11
      0=8000
      1=11025
      2=22050
      3=44100
      4=48000
      5=88200
      6=96000
      7=176400
      8=192000
      9=352800
      10=384000

       
  2. 14 hours ago, Matt M Luthanen said:

    I am troubled that my intention was not clear to everyone - I posted an account of my experience with a Bandlab program*. I wish I had seen such an account before I put so much time into what turned out to be a program suffering from too many malfunctions to safely use. I was not asking for help and the tone of my message was that of caution and nothing else - I clearly explained what I was using, what I was using it for, and what did not work. It was a warning to not try to use the Bandlab DAW program that runs in Chrome for anything but the most simple of projects and then only those you are comfortable with maybe turning out too mangled to salvage.

     

    Matt, the challenge with these kinds of warnings - and the reason sometimes they are met with a little hard looks - is that 99% of the cases the problem is user error.

    To a little more experienced people, it feels a bit like someone driving a car with a manual gearbox, shifting gears without using the clutch, breaking the gearbox and then warning not to use the car because it destroys gears. 

    It may well not be your case: but the way you framed the issue - without any detail, without a hint that you actually are experienced in the software you are using, mixing up two completely different software (a pro-level DAW and a casual-music making web application) suggested an "arrogant beginner" (again, not necessarily true, but the impression you gave).

    That does not warrant hostility, of course - but might explains why more experienced people (who has seen many arrogant beginners) to be a little more curt than it would be nice.

    This is the wrong forum for issues related to the Bandlab webapp, but if you find the right forum and provide some details on what happened and asking what you might have done wrong, you will certainly find people that will give you suggestions. It's one of the ways of learning. It won't help for the project you have already lost, but it may prevent you from losing more - regardless of what software you decide to use.

     

    • Like 3
    • Great Idea 1
  3. Just to give you a heads up, nothing to sweat here. It's fairly common in any production with overdubs that a little bit of leakage is present in mics which may be near to the headphones.. heck, there's artists (and studios) which overdub vocals singing over speakers, inverting the polarity of one of the two and positioning the mics so that the null(s) minimize the leak.. which still ends up being far more substantial of what you hear from headphones. Many people sing with only one cup of the headphones on - and then it's leak galore... 

    In normal circumstances, it's doesn't do any particular harm (so long the leak is reasonably low) - unless you really want to change the backing completely... but then if you do, you'll want to re-record the vocal line as well.

    Use good isolating headphones and enough level for the singer to be comfortable with pitching and following the pitch, and you'll be fine. If you want, you can ask  the artist to place his/her hands on the cups and push them towards the head a little when singing, which seals them even more and makes the leakage almost zero. Many pro artists have learnt to do that in the studio (but then again, many others don't use headphones at all when overdubbing).

     

  4. 12 hours ago, msmcleod said:

    They should then get them to reimplement what they've done in C++, then they'd have a better idea of what goes into programming a DAW ?

    Every time I go back doing something in C++ (last week, it'd been a year since last time) I'm stumped on how the language makes even the simplest thing so incredibly complex and error-prone. What seemed so good in 1992 definitely no longer does! Bit like tape :D Still fast as hell though, so occasionally worth it.

  5. On 4/7/2021 at 5:15 PM, Jacques Boileau said:

    Good point! I just checked and my integrated Intel graphic driver is at the latest...

    Well was worth a try!

    Incidentally, I use the SSLs as well (just got 'em a couple months ago when they were on offer) and they work just fine, so yes, gotta be something specific to your system. Hard to say what. Maybe checking the Event Viewer can give some hints of possible malfunctions?

    • Like 1
  6. 2 hours ago, Glenn Stanton said:

    pretty sure if the simplification simply remove redundant nodes within a given % of the adjacent ones, that would seem straight forward like the video. if you're asking for it to create curves based on the node pattern as showing in the first post, i'd suggest that is way less intuitive for the programming to make - my interpretation of that would have been a bowl not a descending arc to a point... also, i wouldn't want my gaps (the dotted sections) to be automatically filled since i usually mean them to keep the last position steady until next change without needing to consume any automation processing.

    Well, interpolating points with different type of curves is a basic mathematical skill that's been translated into several different, efficient, algorithms a long time ago.

    The hardest bit would be to find a GUI design that allows the user to pick up the points and the curve he thinks fit better in a smooth and effective way.

  7. On 4/4/2021 at 2:01 PM, Nathan Greenly said:

    I work hours on hours to get a good sound with the autotune,reverbs & other effects.

    Don't we all? :)

    On 4/4/2021 at 2:01 PM, Nathan Greenly said:

    I save it because I would like to come back to it another time and I close the project itself and when I reopen the project the autotune doesn't work, and the half the reverbs aren't in the vocals as well, taking my vocals and basically turning them raw when they have the effects saved and everything. just a real pain in the *****. my issue was never the bridging so i have no idea why everyone referred to that I was just saying that I would like to avoid paying for it and that I use bitbridge because there was other forums saying the bridging was the issue.

    If the plugin look loaded and active and the FX bin is active, it's quite odd and it points to something fundamentally wrong with your computer/setup or the project is corrupted. If they look grayed, then they aren't loaded and there's a number of reasons for which that may happen on your system.

     

  8. Trying costs little, but I wouldn't be overly optimistic. A proper VM virtualizes everything, including memory addresses, and adds additional scheduling between the various instances of VMs running on the same physical hardware and while some of the translations happen in hardware, it's definitely not conductive to activities requiring continuous realtime performance. Of course for a sufficiently fast architecture it might work at some point, but it would be costly and inefficient.

    That said, while I have a few different PCs for different rooms, I run the DAW in my main general purpose PC (or vice-versa :) ), and have been doing that for many years without any issue. 

    The trick is to keep the PC "clean" and optimized, understand what steals realtime capabilities (especially when one installs software) and of course know how to fix issues if/when they arise. So a bit depends on your specific knowledge and skills with  computers.

    As an example,  some months ago, when I installed Teams for working with a different business, I did not realize that the wretched thing automatically enrolled my own computer in that business' InTune system.. (due to a bad default by Microsoft, the InTune admins have to explicitly disable enrolling of any computer which access the infrastructure, and mostly don't know so they don't.. with the result that anyone which connects to the organization with Teams get silently enrolled). The results were... funny, and it took a good couple of hours to understand what the problem was and reverse it.

  9. +1 on the multi column menu. I too don't want anything on the screen that it's not Inspector or tracks so never have browser or anything up. I use the per-manufacturer layout as I know what I want (just like I do with my outboard) but certain manufacturers (UAD, I'm looking at you but you're not alone) add everthying even if I have not or never will purchase it and it's pain in the bottom with a long list. I'm in the insert panel, I want to insert something so a menu is great.

    At least the current menu allows to use keys to jump to the effect first letter which eases the annoyance a little, but unfortunately all the UAD plugins start with.. U :D

    • Like 1
  10. 12 hours ago, msmcleod said:

    My studio is way less complicated than it was now that I've repurposed the Yamaha gear.  

    I use an RME Digiface USB ( 4 x ADAT in/out), connected to the outputs of 3 x Fostex VC-8 ADAT converters and the ADAT output of my Focusrite Scarlett 18i20 (which was my main interface until a few months ago).  The Focusrite is set up so I can use it just as a pre for the RME, or I can use it as a main interface if I want. It's also the wordclock master for my whole rig.

    For pres, I've got an Allen & Heath MixWizard WZ3 16:2 with each of the 16 channel outs going to two VC8's, with the ADAT outputs going to the RME.

    The remaining 8 channels are individually switchable between my Alice 828 8 channel outputs, and  4 x TFPro P3's (basically JoeMeek MQ3s) , 2 x Golden Age Pre 73's, and the L/R mix of the Alice - all going into the remaining VC8.  The ADAT output of this VC8 goes to both the RME and the Focusrite.

    The A&H is a really nice clean sounding mixer. The pre's are great and the EQ is pretty transparent - great for shaping sound on the way in, or just leaving it clean.

    The GA Pre 73's are Neve 1073 clones, and the Alice 828 / TFPro P3's are just full of character... they colour the sound in a BIG way... great for some things, but they don't work well on everything.  The P3's can sound great, but are difficult to work with, so I tend to have each one with specific settings for a particular instrument.

    Yeah, was just kidding. My location rig is usually a couple channels of DSP interface (currently a UR28M) so that I can give confort reverb to singers when needed, a few mics I think will work on the session and an old laptop with Cakewalk. For drums in special places, I use either an old Alesis USB 16 mkII with its 8 preamps and a couple of external pres. Either case, the biggest items are the mic stands :D

     

  11. On 3/3/2021 at 12:20 AM, Milton Sica said:

    Hello. I use MLoudnessAnalyzer, but I was doing it at the end of the mix. I tried to make the measurement on each of my tracks, but the processing was quite heavy. Is there a way to measure the necessary gain to be placed on each track and put it directly on the gain fader? Another doubt is that, if so, the final volume, even if the -14 lufs are maintained, will increase. I ask why I have done the metering of -14 lufs in the whole mix and this reduces the volume a lot. Even because the mastering of the bandlab does not have for streaming

     

    Nah - or better you could, but it'd be pointless and  a lot of work for no particular reason. If you really want to concern yourself with LUFS (that's really not a big deal at all, check my post at https://www.theaudioblog.org/post/average-lufs-are-irrelevant), just place your LUFS meter at then end and lower the master fader until you get what you want. But keep in mind (as of the post above) your main problem can be only if the mix is too quiet, not too loud - because some services do not implement gain-up, just gain-down.

    The "reduces the volume a lot" is only because your playback levels are set to listen to CD-mastered music. which peaks at 0dBFS. Turn on the playback volume knob :)

    • Like 1
  12. Hi,

     just a heads up - it's not a big deal and the workaround is obvious and easy, but when changing the buffer size on the RME drivers (v 4.29) to small buffers, (64, 32) Windows gets  a BSOD if Cakewalk is open. No problems in changing it without the DAW open, so the obvious workaround is to close CW before changing the ASIO buffer. It also doesn't seem to be an issue if the buffers are large (from 256 up I seem to remember).

    It's probably as much an issue with the RME driver as with CW but I haven't noticed happening with other applications open, so just thought of mentioning it.

    • Sad 1
  13. 7 hours ago, Shane_B. said:

    I've done the changing of the wrong fader many times too. ?

    I doubt there's any emulation for any of the ART gear. It's low end, but the funny thing is, it's in a lot of pro studio's. I see a lot of youtube video's from guys in real working studio's who love it. Although I see the price is going up on it.

    I use the mechanical bypass buttons on the unit to set the output level so it's equal to the pre-processed input and then I bypass Pipeline in Studio One to A/B the original signal to the ART. If I just rely on the bypass on the unit it's still going through it technically and I'm not getting a true bypass.

    Haha great minds :)

    Yeah the Art is definitely an alright piece of kit. But that's a bit the point... it's never the kit, it's the person using it. Or at most the combination.
    That Neve console with banks of 1073s will still sound like crap in the wrong hands. 

    7 hours ago, Shane_B. said:

    My problem is I get too lost in VST's. I do much better with just a few knobs that I have to manually adjust. Especially with EQ's. I can spend hours tweaking 1 track with Melda's AutoDynamic EQ.

    Absolutely! Haptics and clarity of purpose are the main benefit of outboard. Besides the looks and the fun of course! :DPlus, you usually don't want to buy rackloads of the stuff, so it limits your choice that for some people is very beneficial.

    7 hours ago, Shane_B. said:

    I just compared the remix I did the other night with the ART routing to the original mix of the song completely ITB and there is a night and day difference.

    I don't doubt it. It means you're better at using the ART than whatever else you were using before! 

    On a side note, this can be an indication that you're not yet in terms of goal/solution when it comes to get a mix you like.
    If you were, you would get just as good results ITB as not - because great mixes don't just "happen" - they are the result a very deliberate set of choices which, nowadays, are quite irrelevant of the type of tool - hardware or software. 

    That said, when it comes to learning stuff the sky's the limit and we never stop getting better if we want (it's the fun of playing an instrument or recording or mixing, and most stuff worth doing)... and when you want results here and now, the best tool is the one that gets you there!
     

  14. Usually (not always, but usually) there's simply confirmation bias at work when it comes to outboard. So long you know that you like the ART compressor and and you believe it will add something.. you will hear that something. Just to be clear: I like and use outboard, having  some good pieces of kits. But it's not for the sonic difference - it's more because turning knobs without looking at a screen makes my mixes better-  I hear things, as opposite to looking at them.

    Self-bias is powerful: I once spent minutes fine-tuning a fader and discerning quite a lot of differences with just 0.5 or 1dB differences in fader position... only to realize afterwards that I was manipulating the wrong fader, on a muted channel!   Our mind plays tricks all the time.

    That's also the reason of the DFA button on old consoles. It would be a worthy addition to Cakewalk's excellent console as well. 

    The only way to know if it's true that your HW compressor is adding some magic is to find an emulation, have a friend bounce a mix using the emulation or not (and accounting for any noise etc) and listen without knowing which is which. 

    One of my compressors is a MC77 - a 1176 style FET unit by Purple Audio. It sounds lovely on the right source for the things that a 1176 does, and being even more grabby, it can do things that the original one can't. Once I wanted to use it for a stereo signal - and, not being inclined to buy another for just the one mix, I went and bought the plugin instead. Dialed it in, and went on with it - all good.

    After a few days, having a vocal line to work on, I went to the 77 and had it work its magic. The plugin crossed my mind, and made an experiment, copying the settings on a mono instance. I bounced the two versions and used a little script to scramble names and dates. I could not discern any difference whatsoever. 

    Of course I may have lead ears, but circulated a segment among some friends in the biz, some of whom most definitely do not have lead ears, and they couldn't tell either. Some joked on why the heck I was sending two identical files!

    Some time after I also made experiments between my real UA 610 and the UAD emulation on the Twin, and my real LA2A and the UAD counterpart, with identical result. Take away the knowledge of what's what, and the only magic that's left is your ability to mix.

    Of course I don't mean the compressor doesn't add something good. But it's not it about being physical outboard, it's about the fact that you found a setting you like with a unit you like, easy since you were just turning knobs and there was nothing to look at. That's the big advantage of hardware.

  15. Ah! Literally everything. I am happy to jump from Mozart to 80's Italopop passing via Metallica, Madonna, Herbie Hancock, some blues, Eminem and a detour in country land to end up  with some R&B and hip hop. So long there's a good  melody and some groove I dig it and it inspires me.  The sound of my band and my writing and playing as a songwriter and guitar player is the direct consequence of that.

    Though I've never listened much to current stuff, in any era. I just let the passing of time filter away the 99% of crap and leave the 1% to which is worth listening. So in about 10 years I'll discover some good stuff from 2020 to get inspired from. :D

     

    • Like 2
  16. 22 hours ago, William Burke said:

    As a note.. if you are using the 11rack as an interface... don't. I have it and have used it since around 2012/13 for demos and still do, and love it to bits for that. But as an audio interface, the few months I tried to use it my system was BSODing every week. Talking Windows 7 and Sonar. It had terrible drivers and given Avid total lack of interest in the product, don't think they ever advanced much. 

    My 11rack has been happily connected to the S/PDIF inputs of my (proper) interface for over 7 years and I've been churning demos and put down ideas with zero crashes ever since. Just a heads up.

    • Thanks 1
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