Cristiano Sadun

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.

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.

 

Can be a plugin. Try opening in safe mode, then if it works fine that way,  enable plugins one at a time and see if it's one that causes the problem.

22 hours ago, William Burke said:

Minidump file attached..

_03072020_113551.dmp 2.1 MB · 2 downloads

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.

Bumping the bump. Tempo changes are pretty important for most music and right now it's a pain in the back

On 1/29/2021 at 9:05 PM, jack c. said:

so for this would ya use the daws fader to move down so level for each track is -10 to record?or leave faders at unity gain and use gain knob to achieve this?thanks.jack c.

As others have said, you control the input gain when recording only by the physical controls on the preamp (and by looking at the meters).

That said, I never fade anything during recording, it's much easier to do it afterwards (and you can use both automate the fader or the gain knob, it makes no difference in that context).

Haven't seen a crash in sonar/cakewalk in years. Most crashes I see when helping the people  I tutor have ultimately to do with their interface drivers. The short time I used my 11 rack as interface, I was getting BSODs quite often. Since I switched to a RME interface many years ago, I've not experienced practically any.

It may be , however, that CW is more sensitive to driver issues than other software. The only way that I can consistently make my system crash is if change the interface buffer size to 64 or 32 while CW is open.  When I set the driver so low, I need to do that with CW closed. 

Chromeo...

The piano Roll doesn't look so different !

+1 

I'm not getting any younger - and neither is anybody else! 😀

Just a note. When you are recording to a Dropbox folder, quit Dropbox. Normally it's set to begin copying files to the cloud storage as they are written, and while it often works fine, it will unnecessarily increase the amount of I/O and network communication, which you don't want that when recording.

Once done, you start Dropbox again and it will happily sync everything.

Well truth be told, it wouldn't have to affect the signal path, only the GUI, replacing the double meters with a  single one. 

But: most often there's no reason to use mono interleave on tracks at all. The only exception is about plugins that for some reason treat their  left and right inputs differently; or for some reason one uses a mono plugin (which are a relic from times where CPU resources were scarce). Interleave is about the  plugins - not the source.

Happy to elaborate more if someone wants the full explanation.

+1

The current tempo view is really cumbersome to use, I always end up adding manually tempo changes with the insert new tempo dialog and it'd be oh so much easier to simply have a controlled drawing tool (as opposite to the current mouse-as-pen which affords no precision at all)

This got me curious, so I made a little test - not too much time as I'm in the middle of a big mix, but a big mix is just what we need.

I took two projects: one, the real world mix I'm doing, 24 main tracks,  around 25 buses, around 50 plugins (give or take, didn't count), some of them 32 bit (I love my Variety of Sound stuff) so using BitBridge, several instances of EZDrummer, SI-bass, Ample Bass P, EastWest HBS. The other is a quick recording of a guitar+vocal idea I had yesterday in a break: no plugins, only audio, no Prochannel active

First I closed Cakewalk, and looked at the memory consumption. Then loaded the project, played it, faffed a bit around (nothing substantial, changing the track display from automation to none, removing an inactive plugin etc) and looked at the memory consumption again.  Not much faffing for the simple project of course other than playing it.

I did the whole three times. The working set and private bytes vary a little bit (there can be garbage collecting threads) and I didn't keep the  projects open for long but the experiment gives some indications.

• On the small project there appears to be a small memory leak that can be attributed to cakewalk (since there's really no third party stuff going on).We're talking ome 7-9K on each opening. While this could of course cause problems if you leave CW always open, it's not likely to give much grief for some dozens of open/close operations.
  
  It's also possible that it's cached data and therefore would stabilize if I had run more open/close cycles.
   
• On the large project the leak is bigger, with increased memory consumption in both project open and closed state which are an order of magnitude larger.
  
  Here the culprit has to be either CW or, more likely, one or more of the plugins and synths. Some plugs load samples (EastWest) and since I just bought HBS a month ago, I noticed how it seems a little buggy if you keep opening and closign the project. Bitbridge (and Herbert's 32 bit plugins) seem rock solid.
   
• Data at https://www.dropbox.com/s/nehs7rkx230bmgq/CW%20memory%20consumption.jpg?raw=1
   

My conclusion at this point would be that while CW does indeed seem to have a small memory leak, the culprit of the behaviour you are experiencing is likely to be some plugin or synth you're using.

Something that may have changed from previous version is that CW does more things, so overall the memory required may have increased from Sonar time, so you hit the consequences of problematic plugins more often.

It could, of course, be something to do with CW's plugin loading/unloading as well - hard to say. When I have the time I will make an experiment with only 32 bit plugins (which have shown to unload correctly)  and audio (no synths) to see if we can draw some more conclusions.

2 hours ago, msmcleod said:

http://www.cakewalk.com/Documentation?product=Cakewalk&language=3&help=Tools.04.html

 

Yes, but, it still seems a bug, as it happens randomly when moving a crossfade, i.e. hotspot H with no modifier keys (and the mouse pointer is what it should be, it changes entering the various hotspots, does it?)

It seems to be  more of an issue when there are many takes, more than just the usual two or three. The video refreshes more slowly (I can see the crossfade "painted") and suddenly the hole happens, often not in the active clip whose crossfade is being applied, but at the end of the clip immediately on the right.

When it happens, it tends to happen consistently but I cannot trigger it at will. 

A typical workaround is to create a new clip by entirely overlapping the ones that create the malfunction... the problem never occurs when defining the clip - only when moving the  crossfade between two existing clips.

But the main danger is not noticing that it happens, as then if one saves the project suddenly there's a clip that needs to be recovered from the original audio file, which is a pain in the bottom...

Also, not exactly sure when the issue began to appear, but I guess the last couple updates - I've never seen this issue before.

6 hours ago, reginaldStjohn said:

I haven't used the comping for a while but I have always had issues with the comping system leaving gaps or creating weird crossfades.  I don't have a solution but if can trim your project down to a minimal size you can submit it to the dev team (support@cakewalk.com) and see if they can determine what is causing it.

Thank you, will do. It's odd because for me it's always been working flawlessly - other stuff with take lanes is quirky, especially when pasting in regular clips, but that specific "hole" issue is new to me..

Anybody experiencing problems with the magic tool when comping? (version 26.11.0.098) It's always worked flawlessly but now clip resizing seems to have acquired a bug. 

It happens when I have "cut" a clip with the magic tool in comping mode  (which works fine).

But then when I move the borders of the clip in the track lane (by floating on the crossfade and moving the mouse vertically until it becomes the  "comped clip border moving" tool) and resize the clip with click and drag, CW often shrinks the clip on the right side, leaving a hole between the right side of the clip and the next clip.

It doesn't happen all the time. but quite often. I have no special settings  different than what I've always used (auto crossfade, haven't touched anything related to the arranger etc), nor touching any key modifier when dragging to move the clip border.

If I use the "comp" tool (as opposite to the magic tool in comp mode) all seems to be fine.

Anyone else seeing this?

What BitFlipper writes above is important to understand: you have buffers from and to the interface, and the computer must be able to read/write from/to them fast enough. It might be that your ASIO buffers are too small, but unlikely with a new system. My PC is not super-powerful and I run a 32 samples buffer with 4ms of roundtrip latency with no issues.

Also as BitFlipper says, there are a few cases of hardware components (or more often, their drivers which are not so well written) which generate lots of interrupts - resulting in high DPC count in Windows, slowing down everything else. Certain motherboard-installed network cards from Intel were notorious for this a few years back, especially in laptops, but in recent years  this has been far less of a problem. Sometimes it's surprising, for example the Corsair keyboard driver I had was using an huge amount of CPU for something so trivial as keyboard scanning. 

So excluding that, first of all, your PC must be optimized for real-time processing. That means disabling anything power-management related (both in the BIOS and Windows). removing any bloatware processes and services (your taskbar should be as empty as possible, and as BitFlipper suggests you can use Technet's process explorer to find out if something is overusing CPU). There's an optimization guide from the fellas at Focusrite which is well put together (https://support.focusrite.com/hc/en-gb/articles/207355205-Optimising-your-PC-for-Audio-on-Windows-10).

You want also to ensure the CPU cores are always on (it's an easy tweak in the Windows registry but you can use the free ParkControl utility for example at https://bitsum.com/parkcontrol/ , just turn it off once you've modified the settings).  And of course a SSD beats a rotating hard drive every day of the week and twice on Sunday.

Also, from the msg you show up it seems you have a motherboard sound card. That's likely gonna create I/O problems, so you want to go in Device Manager and disable all the related elements.

Look also at the Windows Services  that are running  - some are not needed and just steal CPU and I/O resources at the wrong moment. 

Best of luck!

19 hours ago, Jacques Boileau said:

This makes so much sense. Every playback system has its own EQ curve. I guess when we hear that mastering needs to make it sound good on all systems, it actually means good on systems that are as flat as the monitoring system used to master and acceptable on all others. I see two main areas in mastering: sound level and EQ. It seems that sound level will translate well over any system and is pretty independent of the playback system, but EQ isn't.

Unfortunatly,  it may then mean that how we EQ in mastering, if it does not translate well on some system, may need to be changed for lesser/different systems. Which means that someone with a high quality/flat system close to ours may never hear how we first intended the song to sound. It's like we are going for the lowest common denominator here...

You really want to preserve the essence of the mix. A way I approach this is that since almost all playback devices worth being called that reproduce the midrange, you really want to make sure that your midrange has a mini-balance that works and keeps the essential elements of the mix intact.. all the rest, is a bonus. I just made a mix where the backing vocals and the tambourines/eggshaker get much less prominent in mono,  but the bass and kick keep being clearly heard even on an old Nokia phone speaker .. and that's alright because some elements are nice to have but aren't essential to the feeling of the song - whereas vox, snare, bass and kick can't be done without. 

So when in mono and on a heavily band-passed speaker, the song folds down nicely to its essence, and still delivers - while if you put it on a great playback system it blossoms with all the bells and whistles.

There's to say that lots of it is really in the arrangement, and it's a progressively harder task to pull off (for me at least) the more complex and dense the arrangement is. Most songs that translate best are - at any given moment - stripped down to a few basic sounds (which change over time of course). 

On 12/13/2020 at 6:49 AM, Craig Anderton said:

And I might add...mastering is a hopeless task. Tonight when doing a final reality check on a master, I listened over four different sets of "pro" headphones, two sets of speakers, and bluetooth in-ears. 

The mixes sounded similar in terms of balance, but the "mastering" sound very different on each one. Yes, I know that a good mastering job is supposed to translate over any system...but it probably won't translate the way you intended. If you can preserve the mix's balance over a zillion different systems committing sonic violence to your music, that's about all you can do.

Yeah.. as I see it, the whole point of a mix is to delivery what  I call an "emotional payload"... the artist wants to convey a certain emotion (and possibly have the listener feel/react in a certain way) and you and the mixing/mastering engineer team wants to make sure that keeps happening over as many different playback situations as possible, from  a  large disco PA to a phone.  And yeah the balance is the most important thing... Say if the way you make the mix has the drums high to get a lot of excitement and energy, you don't want the drums to disappear or be overwhelmed on a mono phone speaker! 

But the sound itself is never gonna be the same , of course 

I'm using a Corsair gaming keyboard and of all things, that driver crashed and started a BSOD cycle. Thanks to Windows RE, I managed to remove it and all is fine. But annoying.

On 12/11/2020 at 9:49 PM, Will_Kaydo said:

In the modern day and age in the digital world, there's a saying amongst those (myself included) that does everything themselves. It goes something like . . . 

MASTER AS YOU MIX So when it comes to the final product - you literally just have to glue everything together and crank that limiter.  There are guys so good at this, they hardly go through an EQ and Compressor at the end - only a limiter. 

This is all purely on personal taste, by what your control room/bedroom, environment, choices and monitors throw at you. 

As Craig, I also think of the two stages as separate, and having as little as possible to do at mastering stage is always my objective, with both my own stuff and client's... but not opening an argument here: in the end of the day, what matters is what comes out of the speakers, not how you get there.

There's also to say that there are (very roughly, of course  ) two classes of mastering engineers - people who  want to/like to/are paid for leave a sonic signature on a mix or an album, and people who don't.  If you give your mix to one of the former type, you will get back something that has, well, his/her sonic signature. If you give one to the  latter, you will get back your mix, as good as it can be without altering the sonics too much. I have the impression that many "names" in mastering (for how much such a thing exists) may be of the first type, but don't know as a fact ..

On 12/10/2020 at 11:42 PM, John said:

That is simply not true. It is truer in the digital world to a very limited degree but not at all true in the time of vinyl records where the term was first used. 

There's nothing I can think of about mastering vinyl that doesn't fit in that category - from how many songs you could fit to choosing the sequence to fit the changing FR of the medium the more you go towards the center (or, of course, realize a compromise with the artist's conceptual vision), to obviously adapting the song to the small dynamic range. All in the name of making stuff sound as good as it could in the medium.  It's not the same of course from a tooling point of view, but that was exactly my point..

Mastering has always been the same, putting on any polish that's necessary to make things sound as good as they can. Which with a very good mix means almost  no polish at all.  Bit like photoshop - you pull the contrast a little, increase or decrease the saturation and so on, so your printed photo looks as great as it can be (which with a poor photo, will still be not so much). If the photo is truly great out of the box, you're just Instagramming it  

The specific ways have of course changed over time - just like once it was to choose the most appropriate paper for prints and now it's reduction of color space..
In audio,  back in the time for example part of that was to make sure it could fit in vinyl's dynamic range, which nowadays that 's no longer an issue  with digital releases.

Still, you need another pair of ears.

That's because what you can ear, you would fix in the mix. If you didn't fix it , it's because you couldn't hear it, for whatever reason. Mastering in the same room and by the same person who's done the mix is pretty pointless (well, you can always splat a limiter of course or these days calculate LUFS and move the master fader..)

As a mixing engineer, you don't need to find someone with special ears: just  competent, and different ears. Their monitoring (room included) must be at least as good as yours.

Personally, I send my mixes to a good mastering engineer, and invariably I can't really hear much difference when they come back, which is as it should be. Exactly because if I could hear the difference, I would have fixed it in the mix. By definition, what the mastering engineer fixes  is something I cannot hear, or don't know how to.

But I do occasionally master for other musicians (not as a job, but as a favor, as my interests lie elsewhere) and the process is nothing complex - mastering is far more "mechanical" in many ways than mixing, even if does require more accurate playback, fresh ears and sometimes a vision for the final product (but only if the mix is "vague", so to say). Just like photoshop!

Sometimes we have exchanged mixes among friends, mastering each other's - just to have fresh ears on someone else's unknown material. It can work very well, with people who know what they're doing.

In these cases my chain is always the same - obviously what links I use and how much. depends from the material.

A couple EQs with a minimum/linear phase option;  meters, scopes and other analysis tools, a multiband compressor with similar low phase shifts at crossovers; an exciter or an harmonics generator; a stereo toolbox, mostly  to mono-ize the low end; occasionally a tape machine emulation. It used to be a limiter, but these days is more often about an integrated LUFS tool  and an attenuator.  And of course for multiple tracks or CDs all the sequencing, breaks and metadata stuff.

Of course desperate mixes may need more, but desperate mixes are best sent back with a list of things to fix. 

On 11/30/2020 at 8:13 PM, John Vere said:

This is the funny thing about coming from the old 8 track days as I still tend to think of my projects as only needing very few tracks to sound good. It scares me to death to read about folks who have 120 tracks on a rock and roll song!  

At most I think my highest track counts are 20 and a lot of that is midi most times. If it's real drums that adds 6 tracks right there, but most of those are still well under 20. 

I'm big on not keeping bad takes. I delete whole tracks if I make more than a few mistakes. I tend to play through the song, delete, re take, delete and over and over until the entire  track is right. You play the track better and better each time anyways.  If I suck that bad I give up and leave it for a better day or come up with a part I can actually play   I do this for everything including midi parts. There's no bad takes lurking in my audio folders.. 

About the only thing I use a extra tracks on is guitar. Bass is always 1.  Vocals will have 1 lead track and then 1 harmony track. I use take lanes sometimes but mostly I do the big delete as I work my way through in segments. I like my vocals to be in sort clips. This is also handy if you find a clunker and resort to Melodyne. Melodyne loves short clips. 

I like to have about 5 or 6  busses,   Drums, Bass, Guitars, Keyboards and midi, and Vocals. Then usually 2 effects busses. Reverb, Delay. 

Well, just a drum kit recorded for a modern sound takes at least 10 tracks for a basic kit.. :D

Often in a recording session you keep absolutely everything, you never know what the artist wants or not  and what may or may not fit.  "Beat it" famously has Michael knocking on a drum case. The session I did for recording bass a few nights ago added some 10 tracks just for it (we took 5 different grooves, each both DI and miked.. and of course they're all in the CW session), each of them has two or three take lanes - I duplicated tracks quickly to more easily differentiate what's what later on.
And even if we have already landed the "right" groove, they'll stay in the session as it may be that at a specifc moment it may be good to have an additional gear to shift.

Actually one thing that annoys me a little in CW is that when you have several tracks with many take lanes (a good singer will need three, a bad singer may need 30) the saving gets very very slow. It doesn't happen if you "just" have many tracks and busses, so it must be some overhead of the take lanes. Haven't tried with the most recent versions though.

As of buses, it's not uncommon to have at least 3 or 4 reverbs to send to, plus for vocals you may have a dedicated reverb, different delays for different sections, parallel compressors,  and of course different processing chains for the different sections. Often there's more buses than tracks!

It's just that the amount of detail in a modern commercial production is staggering - and the better produced they are, the less of the production is visible.. but it's still there.  That's the main difference with homebrew productions (and of course, bad commercial one where the "overproduction" is clearly hearable): the manic attention to detail by someone so skilled that it feels very natural.

If you always go for a very vintage sound and you are the artist so you decide what to throw away immediately you can get on with smaller track counts, but it's a bit of a special situation.


 

Also, you can simply drag the clip down on up keeping Shift+Control pressed, and if you stop outside an existing track, CW will automatically create a new one for you.