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75 workflow improvements to make Cakewalk more intuitive (+ appearance, implementation, etc.)


Olaf

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12 hours ago, msmcleod said:

They've explained it very well in that article.

Unfortunately, that's the Ardour team refusing to implement a feature and using technicalities to excuse it. They do that a lot. If you go to their forums, requesting a feature even for mundane things usually results in something along the lines of "this is not how I (the creator) do things, so there's no reason for this feature to exist." Ask unfa or any YouTuber that uses Ardour and actively tries to help with its development.

EDIT - It's also important to note that Ardour's creator was a ProTools user and essentially replicated what the software does...Down to the lack of stability and bugs. It's also even further behind in features than any other DAW out there, having added some rudimentary polishes to their MIDI stuff in the latest version. The creator is very adamant on keeping the software as close to the way he thinks things should be done and if it differs to that, he won't do it or will convince the rest of the team to not do it. If you push the issue, they'll gladly ignore you or tell you to grab the source code and implement it yourself.

Edited by Bruno de Souza Lino
Adding more relevant points I forgot.
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On 1/16/2021 at 1:51 AM, Kevin Perry said:

Interesting comments on this from the developers of Ardour: https://ardour.org/plugins-in-process.html

Since I already have half a second latency irrespective of buffer size, I wouldn't mind a 7 ms increase to at least solve the crash problems :).

Besides, who of us runs 100+ tracks on a regular basis? I use around 100 plugins, some pretty heavy, but I don't see myself getting to 400 - they don't even fit in my bedroom. I'd say, if you're running that kind of numbers, you're not a home musician, but a pro, so you should (at least afford to) buy a fast computer. So, seriously, that as an argument... Let it crash, cause pros might wanna use it to record on a 386 - and they don't mind the crashes, but they mind buying a new machine. That doesn't fly too high, in my view.

Plus, a 7 ms latency is nothing when mixing - you don't even notice it when you press your Space key, as far as I can tell.

And while tracking, if you record 100 microphones, that's probably an orchestra playing at once, so, again, 7 ms latency is not a concern. It's not even a concern with duplex track-by-track recording, for that number, let alone simultaneous recording.

Besides, there're several DAWs that don't have those problems - and they're pretty fast, too.

So I call bullshit on those explanations.

All that said and aside, getting a little philosophical, I can't see the deep reasoning behind letting a DAW continuously crash until all plugins... become perfect 😶. It's like not getting out of the house until all people become nice. The DAW won't wrok well until all plugin behave.  Is that a valid philosophy 🤨?

How would anyone imagine - with the standards continuously evolving, new technologies always being tested, and new plugins coming out every day - that plugins, generally - let alone all plugins - will ever be perfect? Ever. Like, ever. In this very imperfect world 😪...  Why do all developers come up with version after version after version? Is it because the plugins are perfect? Does that even exist? I'd like to take this opportunity to call for some realism 😛 ...

It's like crashing you car at 150 mph, and saying the problem is not the driving, but the brakes had a bug, the tarmac was an older generation, and the road lines should have been clearer, and the traffic was not properly optimized... If someone told you that, how would you look at that guy in about 20 seconds?

Have you noticed that in CW every user reports a different plugin in the crashes - for me, it's been a few dozens so far, at different stages? So we are to believe that ALL plugins have a crash inducing problem. That's what really turns me off - the lack of logic, it's like Kafka. And, on the other hand, those same plugins don't crash other DAWs - constantly overlooked in these savant considerations, although these two things tell you a lot more than all the savant considerations put together.

As a side comment, the necessity of that article on the Ardour site actually tells me that their users are pretty displeased about it crashing, too. And you can see how much market share Ardour is picking up...

Just thinking, why don't these complains happen in Logic or PT - not that I like those, I'm sure they sound great, but I don't like the graphics, the menus, layout - Logic and Ableton least of all - but just as a standard of performance?

Does PT crash every time a plugin craps out? Would CLA use it if it did? Or do they use some magic plugins? Cause I see them using pretty much all we use. Basic questions. And when, as a developer, you miss all that - pretty much common sense - for savant considerations, it's not seeing the forest for the trees. It has a name, "engineer analysis". Not getting the point, the big picture, or the essentials, but knowing all the formulas. Not to say all engineers are contaminated with it, or that they're not necessary, salutary, and life saving sometimes - especially in this business - but it's a general observation.

Besides, many crashes are not even connected to plugins. What's the explanation when there's no plugin 😉? Tracks that don't play. So on, so forth, I've talked about them so much, there's no point in repeating them.

So the pick up from this "plugin philosophy" is that the DAW will always crash, and there's no intention to make it stop. That's what I see, explanations aside.

It's either fix it or not - that's the bottom line. Knowing why it crashes doesn't make it not crash, and therefore doesn't keep you warm at night. What would help is all this knowledge being actually used for a solution - that would be revolutionary.

Edited by Olaf
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On 1/15/2021 at 4:22 AM, Bruno de Souza Lino said:

With the current setup, I literally have to wait a few seconds before playing back anything after adding an effect or such as that introduces a dropout. Even mundane things, like resetting TBPro's dpMeter cause dropouts. I don't know if it's something with my setup or some configuration in CbB. I don't have those issues in REAPER and Cubase.

Hi, FWIW, I also had an AMD system about 10 years ago and despite latency I did manage to record music with Sonar.  So if you have really strange and long dropouts there could very well be a configuration problem.

After I tried the tips in the links I gave earlier I've noticed a significant improvement, so certainly worth trying. However, there were still some unexplainable latency issues. Once I looked for additional information in order to solve those and tried different suggestions those issues also became acceptable.

Here are the  latency issues + suggestions:

NDIS.SYS:

  1. AMD? In the device manager, find "ATA / ATAPI IDE Controllers", select "AMD Sata controller" and disable it.
  2. Disable Driver Verifier:
    Open an elevated Command Prompt by right-clicking on CMD.EXE shortcut in your start menu and selecting Run As Administrator from the context menu.
    Type the following command:
    Code:
    VERIFIER /RESET
    Reboot the computer
    Check if the issues still persist and, in case they do, generate a new trace
  3. Disable AV or any program that filters internet packets

TCPIP.SYS:

  1. Run TCPOptimizer
  2. Uninstall Bonjour.
  3. If you have a third party AV with firewall: disable "Windows Defender Firewall" service. Even if its turned off in Control Panel, the "Windows Defender Firewall" service (MpsSvc) can still be running in the background and causing issues. Completely disable the service via the registry and reboot.  HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\MpsSvc
    Change "Start" from "2" to "4"
    Reboot the PC

DXGKRNL.SYS

  1. Set NVIDIA Surround, PhysX to GPU, not CPU
  2. Check if an IRQ is shared between GPU and something else like USB. Check internet for ways to change IRQs. Tip: If you no longer use legacy ports, disable them in the system BIOS.

Hope this helps

 

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21 hours ago, Olaf said:

I like the one about emulation and marketing hype. How many CLA Epic emails I keep receiving again?

 

13 hours ago, Teegarden said:

Hi, FWIW, I also had an AMD system about 10 years ago and despite latency I did manage to record music with Sonar.  So if you have really strange and long dropouts there could very well be a configuration problem.

After I tried the tips in the links I gave earlier I've noticed a significant improvement, so certainly worth trying. However, there were still some unexplainable latency issues. Once I looked for additional information in order to solve those and tried different suggestions those issues also became acceptable.

Here are the  latency issues + suggestions:

NDIS.SYS:

  1. AMD? In the device manager, find "ATA / ATAPI IDE Controllers", select "AMD Sata controller" and disable it.
  2. Disable Driver Verifier:
    Open an elevated Command Prompt by right-clicking on CMD.EXE shortcut in your start menu and selecting Run As Administrator from the context menu.
    Type the following command:
    Code:
    VERIFIER /RESET
    Reboot the computer
    Check if the issues still persist and, in case they do, generate a new trace
  3. Disable AV or any program that filters internet packets

TCPIP.SYS:

  1. Run TCPOptimizer
  2. Uninstall Bonjour.
  3. If you have a third party AV with firewall: disable "Windows Defender Firewall" service. Even if its turned off in Control Panel, the "Windows Defender Firewall" service (MpsSvc) can still be running in the background and causing issues. Completely disable the service via the registry and reboot.  HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\MpsSvc
    Change "Start" from "2" to "4"
    Reboot the PC

DXGKRNL.SYS

  1. Set NVIDIA Surround, PhysX to GPU, not CPU
  2. Check if an IRQ is shared between GPU and something else like USB. Check internet for ways to change IRQs. Tip: If you no longer use legacy ports, disable them in the system BIOS.

Hope this helps

 

Ironically, my problem is not latency. I can record and do quite a bit on 256 samples but somewhere along the way, CbB starts to struggle rendering some graphical stuff when there's a lot of them on screen. Reaper does the same thing but only the UI becomes slow. No dropouts happen.

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1 hour ago, Bruno de Souza Lino said:

Ironically, my problem is not latency. I can record and do quite a bit on 256 samples but somewhere along the way, CbB starts to struggle rendering some graphical stuff when there's a lot of them on screen. Reaper does the same thing but only the UI becomes slow. No dropouts happen.

Seems like a GPU related problem. I understood that the graphical interface of CbB could be improved with scalable graphics etc. Maybe the current state is heavier on the GPU side, which is too much for some older systems . BTW, sometimes certain graphics processes are done by the CPU. If that's the case I would change those to the GPU if possible.

Did you try changing the GPU? Did you run LatencyMon for a while to check which process could be the culprit? Did you check Windows Reliability Monitor to see what crashed? IRQ sharing of the graphics card with something else? 

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Just now, Teegarden said:

Seems like a GPU related problem. I understood that the graphical interface of CbB could be improved with scalable graphics etc. Maybe the current state is heavier on the GPU side, which is too much for some older systems . BTW, sometimes certain graphics processes are done by the CPU. If that's the case I would change those to the GPU if possible.

Did you try changing the GPU? Did you run LatencyMon for a while to check which process could be the culprit? Did you check Windows Reliability Monitor to see what crashed? IRQ sharing of the graphics card with something else? 

I know. Don't have resources for that. All the money I had bought an audio interface. I wasn't willing to guess if the problem with the gpu I bought in the past not working was the power supply, as the power supply would cost all the money I had at the time.

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15 hours ago, Teegarden said:

Hi, FWIW, I also had an AMD system about 10 years ago and despite latency I did manage to record music with Sonar.  So if you have really strange and long dropouts there could very well be a configuration problem.

After I tried the tips in the links I gave earlier I've noticed a significant improvement, so certainly worth trying. However, there were still some unexplainable latency issues. Once I looked for additional information in order to solve those and tried different suggestions those issues also became acceptable.

Here are the  latency issues + suggestions:

NDIS.SYS:

  1. AMD? In the device manager, find "ATA / ATAPI IDE Controllers", select "AMD Sata controller" and disable it.
  2. Disable Driver Verifier:
    Open an elevated Command Prompt by right-clicking on CMD.EXE shortcut in your start menu and selecting Run As Administrator from the context menu.
    Type the following command:
    Code:
    VERIFIER /RESET
    Reboot the computer
    Check if the issues still persist and, in case they do, generate a new trace
  3. Disable AV or any program that filters internet packets

TCPIP.SYS:

  1. Run TCPOptimizer
  2. Uninstall Bonjour.
  3. If you have a third party AV with firewall: disable "Windows Defender Firewall" service. Even if its turned off in Control Panel, the "Windows Defender Firewall" service (MpsSvc) can still be running in the background and causing issues. Completely disable the service via the registry and reboot.  HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\MpsSvc
    Change "Start" from "2" to "4"
    Reboot the PC

DXGKRNL.SYS

  1. Set NVIDIA Surround, PhysX to GPU, not CPU
  2. Check if an IRQ is shared between GPU and something else like USB. Check internet for ways to change IRQs. Tip: If you no longer use legacy ports, disable them in the system BIOS.

Hope this helps

Good advice.

 

1 hour ago, Bruno de Souza Lino said:

I like the one about emulation and marketing hype. How many CLA Epic emails I keep receiving again?

 

Ironically, my problem is not latency. I can record and do quite a bit on 256 samples but somewhere along the way, CbB starts to struggle rendering some graphical stuff when there's a lot of them on screen. Reaper does the same thing but only the UI becomes slow. No dropouts happen.

 

I've been saying for a while now - including to the developers - that a lot of the problems CW has seem to be related to the way the engine interacts with the Windows graphic processing, and the way it interacts with audio processing - since a lot of the audio problems seemed to also have a visualization related component - and how updates overwrite the registry - stability wise. Can't tell how or what, I'm not an engineer, but your problem seems to confirm that.

 

1 hour ago, Bruno de Souza Lino said:

I like the one about emulation and marketing hype. How many CLA Epic emails I keep receiving again?

 

Actually, I agree with that one. I can say from experience that analog emulations make a huge difference - if you're into the response they have - and I'm a huge fan. More weight, more depth, width, smoother transients, less mud, more punch - all that you get just by running sound through a good emulation - separation, if there's summing involved - without actively doing any processing. But you need to pay attention to gain stage your signal right, that can make a huge improvement, or on the contrary, muddy things up, distort or mellow the attack, etc. - just they way analog hardware works.

The problem Epic has, in my view, is that all the delays in that "palette" are tape, so they sound relatively the same - which makes for not much of a palette, delay wise. Plus, it's rather heavy on resources. Someone measured a single instance at about 2.5% CPU - don't know what setup he had. But I like the concept, and the sound is high quality - if it had more variation in the delays, and in the dampening times of the reverbs, it could become my go to for effects - very convenient for choosing the sound - and also building combined effects.

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4 hours ago, Olaf said:

Actually, I agree with that one. I can say from experience that analog emulations make a huge difference - if you're into the response they have - and I'm a huge fan. More weight, more depth, width, smoother transients, less mud, more punch - all that you get just by running sound through a good emulation - separation, if there's summing involved - without actively doing any processing. But you need to pay attention to gain stage your signal right, that can make a huge improvement, or on the contrary, muddy things up, distort or mellow the attack, etc. - just they way analog hardware works.

The problem Epic has, in my view, is that all the delays in that "palette" are tape, so they sound relatively the same - which makes for not much of a palette, delay wise. Plus, it's rather heavy on resources. Someone measured a single instance at about 2.5% CPU - don't know what setup he had. But I like the concept, and the sound is high quality - if it had more variation in the delays, and in the dampening times of the reverbs, it could become my go to for effects - very convenient for choosing the sound - and also building combined effects.

I've never used analog units and I entered into making music in 2002, when people were recording and mixing in the box already. I don't believe there's any defect present in analog units that cannot be replicated digitally. And you have determinism, which means all defects can be repeatable. I'm referring to those tonal characteristics as "defects" because of a few things. Firstly, when you have to record into tape, it has noise you can hear and each batch sounds different, so you have to EQ to the tape you're using and print some things very hot to overcome noise. When digital first came out, engineers thought it was like tape and recorded and mixed to it like they did with tape. This is the reason why the early digital audio catalog sounds harsh and "digital".

I have tried to use Ecosphere, which was the plugin Waves have away last year. I don't remember how it sounded, but I remember several people mentioning how similar it was to the Lexicon 480L and that you could learn how to use the plugin by reading the 480L manual. What makes Waves plugins heavy on resources is their copy protection scheme. The reason their plugins take longer to load the first time you load them is because Waves starts a server process on your machine with the some purpose of loading and dealing with its plugins on top of any extra processing the plugin itself does. Plugins I tried to use but they would cause dropouts by simply being loaded were the PRS Amps and Abbey Road Chambers. I don't even have to run sound through them.

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5 hours ago, Bruno de Souza Lino said:

I've never used analog units and I entered into making music in 2002, when people were recording and mixing in the box already. I don't believe there's any defect present in analog units that cannot be replicated digitally. And you have determinism, which means all defects can be repeatable. I'm referring to those tonal characteristics as "defects" because of a few things. Firstly, when you have to record into tape, it has noise you can hear and each batch sounds different, so you have to EQ to the tape you're using and print some things very hot to overcome noise. When digital first came out, engineers thought it was like tape and recorded and mixed to it like they did with tape. This is the reason why the early digital audio catalog sounds harsh and "digital".

I have tried to use Ecosphere, which was the plugin Waves have away last year. I don't remember how it sounded, but I remember several people mentioning how similar it was to the Lexicon 480L and that you could learn how to use the plugin by reading the 480L manual. What makes Waves plugins heavy on resources is their copy protection scheme. The reason their plugins take longer to load the first time you load them is because Waves starts a server process on your machine with the some purpose of loading and dealing with its plugins on top of any extra processing the plugin itself does. Plugins I tried to use but they would cause dropouts by simply being loaded were the PRS Amps and Abbey Road Chambers. I don't even have to run sound through them.

Digital recordings sounds thin, flat and harsh by default, regardless of the input volume, unless you warm them up. The input gain doesn't have much of an effect on that, at most it can clip and stutter. The reason why they sound harsh is exactly that they don't have those "defects" which correspond to - and have been optimized, for decades, to match - our natural hearing preferences - and the way sound naturally behaves in a physical environment.

Also, in my experience you can't hear tape noise - unless you amplify the playback through the roof and/or you use very worn-out tapes - even then, you get more wobble and high end rolloff, before you actually hear the noise. You can feel tape noise, now that you are able to make the comparison, but not really hear it - on a good recording, anyway.

I have recorded on cassette tapes, which were a lot lower in quality than reel tapes, and could never hear the noise, if I had a good input volume going in, and didn't try to break the volume knob afterwards. And I've listened to thousands of reel tapes and cassettes, many recorded from a third-fourth hand copy, and I was never bothered by noise - except for very old tapes or very poor copies.

To hear the noise, you'd have to crank the volume so much that you'd have to get out of the room when the playback actually started - and maybe protect your windows, too.

Fe oxyde formulations were better, noise wise, than Cr, which were hissier - favoring hyped highs - but even with those, for a new tape recorded with a high input, you wouldn't hear it.

It's not hard to check for yourself - do a simple test: take a song recorded on tape and listen to it 😳. Crank it up, wait for a moment of silence/low instrumentation - even then you won't hear anything. Maybe on "I'm not in love", but that's an intentional effect.

It wasn't that bad. It wasn't cave man recording, and the world wasn't invented in the 2000, as many - who usually don't know anything about anything before the year 2000 - seem to believe. The problem with tape appeared when you had to comp, bounce and transfer repeatedly, that's where you could actually compound the noise to become a problem. But even then, on good machines, with good tape, there was hassle, but it was actually doable - as proven by millions of recordings.

These are just things people say on the web, repeating stuff they, many times, have no idea about, based on misunderstood theoretical considerations made "in-vitro".

People - back then - thought noise was bad altogether because they had to be concerned with not getting too much of it, and they had no way of knowing how things would sound without it. The 90s, however, gave us the answer to that one: reaaaally bad.

Noise, in itself, is not something to avoid - TOO MUCH noise is. Again, that's in-vitro prejudice resulting from theoretical considerations overamplifying the concern of avoiding getting too much noise. On the contrary, in the right amounts, noise has a beautiful effect. Try it for yourself, it's a lot better than taking web comments for granted.

It adds punch, depth and glue, it smoothens out the transients - exactly that harshness - it's a big part of it. It's not just "saturation", it's also in-built EQ curves, it's gentle compression, there's a ton of things going on. I think most people who worked in the analog era prefer recording to tape to this day - if they have one available, bar the technical hassle - analog preamps, which are noisier, etc. If not, they at least use emulations.

To me, not just recordings, but even digitally generated sound sounds flat and lifeless, unless you add noise, non-linearities, saturation, so on, to it. It doesn't have the same weight, punch, all the rest.

The reverb in Echosphere emulates a Lexicon plate algorithm, from what Waves say, but I don't think the Lexicon manual will help you much, since the plugin doesn't have the same controls - besides, the controls are so simple, that the manual can only waste you time.

But I think it will help you a lot more in making assessments if you remembered how it sounds rather than what somebody on Youtube said about it.

Edited by Olaf
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1 minute ago, Olaf said:

Digital recordings sounds thin, flat and harsh by default, regardless of the input volume, unless you warm them up. The input gain doesn't have much of an effect on that, at most it can clip and stutter. The reason why they sound harsh is exactly that they don't have those "defects" which correspond to - and have been optimized, for decades, to match - our natural hearing preferences - and the way sound naturally behaves in a physical environment.

The reason you think that is the case is because digital has more dynamic range and frequencies. Remember that vinyl and tape and band limited and you have to print fairly hot into them because they have audible noise, which doesn't happen in digital. As Bobby Clearmountain mentions, you have people that think recording to digital is like using tape and people that think that recording analog can be done flat like you do in digital. The early digital recordings sound harsh because engineers thought it was like tape and would push it like you have to do with tape in order to overcome the noise it has. You'll also see modern tape recordings with no high end because people using those don't know you have to EQ to the tape you're using.

6 minutes ago, Olaf said:

Also, in my experience you can't hear tape noise - unless you amplify the playback through the roof and/or you use very worn-out tapes - even then, you get more wobble and high end rolloff, before you actually hear the noise. You can feel tape noise, now that you are able to make the comparison, but not really hear it - on a good recording, anyway.

You can't hear tape noise on modern machines. No two tape machines or tape reels sound the same or have the same tonal characteristics.

7 minutes ago, Olaf said:

I have recorded on cassette tapes, which were a lot lower in quality than reel tapes, and could never hear the noise, if I had a good input volume going in, and didn't try to break the volume knob afterwards. And I've listened to thousands of reel tapes and cassettes, many recorded from a third-fourth hand copy, and I was never bothered by noise - except for very old tapes or very poor copies.

I can hear hiss on tapes without even having to measure it. Also consider that some decks have Dolby noise reduction, which reduces the hiss on tape. But it's there.

8 minutes ago, Olaf said:

To hear the noise, you'd have to crank the volume so much that you'd have to get out of the room when the playback actually started - and maybe protect your windows, too.

On digital, you can only hear the noise if you turn the audio up to the point where you could lose your hearing permanently. Modern DACs have noise 150 dB down the signal. A 120 dB signal already causes temporary hearing loss. Two orders of magnitude lower than tape.

 

11 minutes ago, Olaf said:

It wasn't that bad. It wasn't cave man recording, and the world wasn't invented in the 2000, as many - who usually don't know anything about anything before the year 2000 - seem to believe. The problem with tape appeared when you had to comp, bounce and transfer repeatedly, that's where you could actually compound the noise to become a problem. But even then, on good machines, with good tape, there was hassle, but it was actually doable - as proven by millions of recordings.

These are just things people say on the web, repeating stuff they, many times, have no idea about, based on misunderstood theoretical considerations made "in-vitro".

People - back then - thought noise was bad altogether because they had to be concerned with not getting too much of it, and they had no way of knowing how things would sound without it. The 90s, however, gave us the answer to that one: reaaaally bad.

Noise, in itself, is not something to avoid - TOO MUCH noise is. Again, that's in-vitro prejudice resulting from theoretical considerations overamplifying the concern of avoiding getting too much noise. On the contrary, in the right amounts, noise has a beautiful effect. Try it for yourself, it's a lot better than taking web comments for granted.

Let's not forget that the first recording done fully in the box was in 1999. You could argue that we still use caveman technology in digital, except software doesn't have many of the physical limitations analog gear has. While you can make digital sound analog, it's a very large challenge to make analog soung digital.

15 minutes ago, Olaf said:

I think most people who worked in the analog era prefer recording to tape to this day - if they have one available, bar the technical hassle. If not, they at least use tape emulations.

Why wouldn't you use tape emulation? You have the defective band limited sound with none of the drawbacks or can add them in.

18 minutes ago, Olaf said:

The reverb in Echosphere emulates a Lexicon plate algorithm, from what Waves say, but I don't think the Lexicon manual will help you much, since the plugin doesn't have the same controls - besides, the controls are so simple, that the manual can only waste you time.

But I think it will help you a lot more in making assessments if you remembered how it sounds rather than what somebody on Youtube said about it.

I don't use Waves plugins anymore. Their setup is convoluted and complicated and Echosphere doesn't work on Windows 7. There's also the fact that none of them have PDC nor work well with PDC. You have to go to their website and check how many ms of latency each plugin adds to the signal.

Once again, I'm not arguing the desirable artifacts and coloration analog gear gives you. What is demonstrably wrong is the notion that digital cannot emulate it. When you take our limited hearing into consideration, digital has infinite headroom and sound quality. It reproduces every frequency we can hear exactly like we hear it recorded through microphones up to the loudest sound we can hear. Analog can't do it. Also, no two pieces of analog gear sound exactly the same either. Electrical components have tolerances, their performance changes with temperature and humidity, as well as voltage and current fluctuations...And so on. Digital is deterministic. A certain input will give you the same output every single time.

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I haven't said digital can't emulate analog, it's already past evidence that it can, and in one software generation I don't think you'll be able to tell the difference for the better makers out there.

Analog can't do digital. Sure. So? We don't want digital, anyway - in terms of sound. Convenience wise, sure, who wouldn't? Soundwise... nah. So, if you have no problem committing to a sound, how is not getting digital sound bad? That's not what you want to get, anyway.

I think calling it "defective" really restricts understanding and prejudices your approach and your assessment on analog. Don't get me wrong, I don't want to go back to pulling out the dozen of cables, and moving things around, and bla-bla. I don't fetishize analog. But the sound is unbeatable. Because here's the thing - the gear that you used, even low quality, wasn't just any old analog gear - you wouldn't record on a cuckoo clock :) - but the one that - by accident or craft, or both - was optimized to sound great to our ears.

So "defective" in relation to what? To a "pristine" flat recording? Sure, but that doesn't exist. EVERYTHING is "defective" in relation to a perfectly flat recording - which is an absolutely useless theoretical consideration, more of an ideal referential than anything. The question is not whether a recording is flat or not - which, again, now that we can compare, is actually bad, as a final sound - of course you'd want the input to be flat, to allow yourself options in "unflatness" later, as a cheatcode, but not the end result - but rather, does it sound good or not? From that perspective, what can label analog as "defective", instead of "beautifully optimized" for instance - even if by accident or limitation, in a number of features? It's only "defective" from a useless theoretical perspective no one cares about, when you listen. Flat eq curves is white noise. You don't want that. Once you agree your signal shouldn't look like that, the rest is just choosing which curve you like best. It's as easy as that.

It's like "efficiency" - you always need to ask yourself "efficiency in relation to what" to make sure you're not doing more harm in the name of a word. Different topic, anyway, but I don't like sterile, unapplied theoretical considerations - they make for the biggest judgment and logical mistakes ever.

Again, harshness has nothing to do with the volume. It's digital, it doesn't change tone according to gain. That's not what Bob Clearmountain was saying. He was saying engineers - this is a good example of the classic "apply formulas instead of common sense" that engineers sometimes excel at - were pushing the treble, in digital, the way they did on tape. You see that to this day, on all kinds of tutorials, and you wonder "Jesus, don't these guys have ears?". They call it "air". That's not air. That's horrid. That's screeching. Air is pushing the treble on the room reverberations that you'd have embedded in a recording and that would tame the highs of the main, anyway - in addition to the taming the tape and preamps did. That's why they called it air.

But regardless, even if you don't, it still sounds harsh. Try it. I record guitars. Bass. I have once recorded drums. They always do. They have a fizz. Here's the thing, guitar amps naturally sound a little harsh, so they need to be tamed - in that regard, tape was just a match made in heaven. But more than harsh, they sound flat and lifeless, without vibrance. And if you use amps sims, like a I do, you'd cringe without warming.

Watch this, I loved it. All tracks are obviously tracked to tape - listen for any noise, but before anything listen to how they sound just by being run through the gear, before any active processing. Listen to those hats. Listen to the bass. And it's a J console, which is supposed to be a lot cleaner than an E, for instance.

Too bad you don't like Waves, I'm a big fan of the CLA Mixhub, I'd recommend it to anybody. New generation stuff sounds amazing. I've made my first synth sounds using Element 2.

In relation to digital signal damaging your ears before you can hear noise, that's what I was trying to tell you about analog - it's the same. You couldn't hear the noise until you amplified to insane levels - again, on a good recording, good tape, etc. The noise was well below hearing levels.

Anyway, gotta sign out for the night. We can resume. Have a great evening!

 

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On 1/16/2021 at 1:52 AM, Bruno de Souza Lino said:

Unfortunately, that's the Ardour team refusing to implement a feature and using technicalities to excuse it. They do that a lot. If you go to their forums, requesting a feature even for mundane things usually results in something along the lines of "this is not how I (the creator) do things, so there's no reason for this feature to exist." Ask unfa or any YouTuber that uses Ardour and actively tries to help with its development.

EDIT - It's also important to note that Ardour's creator was a ProTools user and essentially replicated what the software does...Down to the lack of stability and bugs. It's also even further behind in features than any other DAW out there, having added some rudimentary polishes to their MIDI stuff in the latest version. The creator is very adamant on keeping the software as close to the way he thinks things should be done and if it differs to that, he won't do it or will convince the rest of the team to not do it. If you push the issue, they'll gladly ignore you or tell you to grab the source code and implement it yourself.

I do agree it's over the top (you'd not necessarily want to sandbox every single plugin!), but the principle is correct, no?  If not, then where is it wrong?

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10 hours ago, Olaf said:

Analog can't do digital. Sure. So? We don't want digital, anyway - in terms of sound. Convenience wise, sure, who wouldn't? Soundwise... nah. So, if you have no problem committing to a sound, how is not getting digital sound bad? That's not what you want to get, anyway.

I think the following video gives some nice insights on this topic: Why Are Synths So Difficult To Mix???

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11 hours ago, Olaf said:

I haven't said digital can't emulate analog, it's already past evidence that it can, and in one software generation I don't think you'll be able to tell the difference for the better makers out there.

Analog can't do digital. Sure. So? We don't want digital, anyway - in terms of sound. Convenience wise, sure, who wouldn't? Soundwise... nah. So, if you have no problem committing to a sound, how is not getting digital sound bad? That's not what you want to get, anyway.

I think calling it "defective" really restricts understanding and prejudices your approach and your assessment on analog. Don't get me wrong, I don't want to go back to pulling out the dozen of cables, and moving things around, and bla-bla. I don't fetishize analog. But the sound is unbeatable. Because here's the thing - the gear that you used, even low quality, wasn't just any old analog gear - you wouldn't record on a cuckoo clock :) - but the one that - by accident or craft, or both - was optimized to sound great to our ears.

What people that fetishize analog gear don't seem to understand is that those pieces of gear were built the way they are and sound the way they are because that was the technology they had at the time. The many issues you see in analog gear are a compromise they had to take in because of technological and physical limitations. Engineers of that time would've killed to have the crystal clear sound we have nowadays.

11 hours ago, Olaf said:

Again, harshness has nothing to do with the volume. It's digital, it doesn't change tone according to gain. That's not what Bob Clearmountain was saying. He was saying engineers - this is a good example of the classic "apply formulas instead of common sense" that engineers sometimes excel at - were pushing the treble, in digital, the way they did on tape. You see that to this day, on all kinds of tutorials, and you wonder "Jesus, don't these guys have ears?". They call it "air". That's not air. That's horrid. That's screeching. Air is pushing the treble on the room reverberations that you'd have embedded in a recording and that would tame the highs of the main, anyway - in addition to the taming the tape and preamps did. That's why they called it air.

I never said it did. And by constantly going into this whole "theoretical" vs "practical" tirade, you're doing exactly what Bob is taking about. "Digital songs harsh because I prefer the analog sound." If you don't know how to make digital sound good, that's not a problem with the sound.

It's your opinion and it's okay. It's okay to be wrong.

11 hours ago, Olaf said:

In relation to digital signal damaging your ears before you can hear noise, that's what I was trying to tell you about analog - it's the same. You couldn't hear the noise until you amplified to insane levels - again, on a good recording, good tape, etc. The noise was well below hearing levels.

So, how you're proving analog gear and tape has less noise by showing a video from last year? You don't know from where that audio is playing, if that console has been reconditioned with modern components... That's like a girl coming to you stating you impregnated her and only showing her belly as a proof of that.

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23 hours ago, Teegarden said:

I think the following video gives some nice insights on this topic: Why Are Synths So Difficult To Mix???

Beautiful. Actually cool to watch, who would have said the production actually had a great sense of humor, too. Very British, reminded me of Droopy - that little dog that had a long face when he said he was happy.

The near/far - whisper/screaming examples are an illustration that I use when speaking about the the necessity to pass every dry sound/sample through a room emulation/IR, to recreate the natural behavior of sound playing in a space, without which to me it sounds flat, and artificial. It's a big part of generating a three-dimensionality that's missing in digital mixes, it helps you define the size of every instrument, it helps you organize the sounds, depth wise, and it also contributes to reducing harshness even while actually livening up a sound - in connection to the mention about "air", before. And it has a huge role in defining a mix's character. Another thing that's missing in today's everyday digital mixes, even though all the tools are there - because of some false equations developed in consequence of attaching ego to labels like "modern", etc., and not putting in the hours to listen to great music from the past.

It's the first time I've ever seen something like that delay loop cassette. A lot of high level thinking and precise engineering went into the electro-mechanical devices of those days. My hi-fi stereo with autoreverse on remote, loop playing, tape counter, and CD changer, with programmable recording from a multi- CD customizable playlist. Lots of pressure sensors, mechanical arms, gears... Even a simple VCR. Everyday consumer products - a lot of complex engineering went into those, with very little room for error.

Thanks for the video.

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Glad you liked it.
It think that everyone who's using a DAW should at least see a video like this once to really understand what he's doing. I'm al the time trying to get mixes sound as analog as possible, just because it sound way more pleasant and especially less exhausting to my ears...
And then there's the virtually unlimited number of ways you can use that to create different sounds, fitting different styles.

Regarding music from the past, todays vocals are virtually perfect and regularly slightly electronic thanks to Antares and alike. Recordings from the sixties and seventies don't sound perfect, are not always perfectly timed and are sometimes even false, but at least to me they sound warmer with lots of character and more emotion, exactly what I'm missing from todays recordings.  And just like you say about the every day consumer products, in the analog studies there's been tons of development to get sounds perfected. No wonder there are so many plugin developers trying to get introduce the "latest, best emulation" of the renown analog hardware consoles and FX processors, including tape saturation.

 

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23 hours ago, Bruno de Souza Lino said:

What people that fetishize analog gear don't seem to understand is that those pieces of gear were built the way they are and sound the way they are because that was the technology they had at the time. The many issues you see in analog gear are a compromise they had to take in because of technological and physical limitations. Engineers of that time would've killed to have the crystal clear sound we have nowadays.

Actually they wouldn't, as it turns out, it's ironical, isn't it ;)? They thought they did until they made the comparison - I've already answered that - and realized you can actually sound even worse than you did with distortion and noise - you can sound "digital", that is. That's why they invented the term. And you can clearly see it, in the ton of analog emulations, noise and saturation plugins that they use now.

It's not hard to see it everywhere, beyond blank in-vitro considerations without applicability, like "compromise", "limitations". What you may fail to see is that they overcame those limitations, in a way that works best with our ears, whereas all you have to offer for the lack of good sound is sterile considerations that actually come from a distorted theoretical perspective - 0 Kelvin - a so called neutral point that no one can use or needs.

23 hours ago, Bruno de Souza Lino said:

"Digital songs harsh because I prefer the analog sound."

No, digital sounds harsh because it sounds harsh - and that's why I prefer analog sound. That's the correct logical sequence. There's no "fetish" for analog sound that you're freely and baselessly attributing to people that you don't know. By the way, great way to turn a conversation on principle into personal attacks. Do I sense you becoming defensive? Hint: when you feel you do, it might be a sign you need to rethink the things you're defensive about.

If anything I could say I see a fetish for digital from those using uninsightful and sterile considerations that they attach their ego to, instead of their ears.

While there is a margin for personal interpretation and taste, obviously, you can't "subjective" your way out of everything - while the speaker still sounds the way it does. There's also a right and a wrong, and it pertains to the natural quality of a sound, and the timbre of the source.

For instance this below is not "clear", it's harsh - and it comes from a bad instrumental mix.

And it's easy to tell the difference, beyond the bullshit: ask yourself if you've ever heard a human voice, anywhere, sounding like that. While we improve clarity, hype-up and change the natural sound all the time, apply cheat-codes - it's what we do - there's a margin for that before you lose touch, and things actually start to sound bad.

 

23 hours ago, Bruno de Souza Lino said:

If you don't know how to make digital sound good, that's not a problem with the sound.

I'm not gonna go into personal considerations of what I can and can't do 😉, it suffices to say that this very sentence should have made you think, if you had the inclination - it's a sentence that actually says analog sound is better, if you took the time to take it in.

With good sounding analog you don't need to make anything sound good, it just did - obviously in relation to the source - by running it through the gear. What you would do is make it sound different, as needed. That's actually a good sign to recognize good gear, you can't make it sound bad - you can make it sound different in many ways, but not bad, tone wise - and you don't need to "make it" sound good - it already does. That should tell you a lot right there.

Anyway, I'm withdrawing from the conversation, don't feel like engaging in a personal to and fro. Suffice to say that, for all the incredible tools that we have these days, I've never seen a larger amount of bad sounding mixes on commercial radio than these days - and it seems to be the norm. That's another thing that should tell you a lot. And they come from sterile considerations like that, that miss the point from an attachment to words.

 

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1 hour ago, Teegarden said:

It think that everyone who's using a DAW should at least see a video like this once to really understand what he's doing.

Absolutely agree, but I think many have a defense mechanism against doing that - that is attached to finding out they're wrong and/or they're not necessarily the bee's knees just for the amazing merit of being born in the 2000s/90s. It's the same thing as the censorship and intolerance on the Left - they just can't find out they're wrong, because they have their ego attached to it, and their identity - in the absence of another.

1 hour ago, Teegarden said:

it sound way more pleasant and especially less exhausting to my ears...

Yeah, exactly.

1 hour ago, Teegarden said:

Regarding music from the past, todays vocals are virtually perfect and regularly slightly electronic thanks to Antares and alike. Recordings from the sixties and seventies don't sound perfect, are not always perfectly timed and are sometimes even false, but at least to me they sound warmer with lots of character and more emotion, exactly what I'm missing from todays recordings.

They are smoother and more polished than ever before, that's true, but to me they're far from being perfect. That more unnatural than ever before, too. You can hear any lisp, plop, lip sticking, it's like the singer's got their mouth right in your ear, and it's very unpleasant, and aggressive. That's not how you hear people. Add to that the dryness of the sound which gives it a flatness and is, again, unnatural, and the, many times, exaggerated highs, and it's... uugghh.

I agree with you 100% on the performance. It's amazing to me - and this is something those of the younger generations who don't spend the time listening, cultivating themselves - instead they just prefer projecting ego - and understanding what they see/hear, the level of mastery you needed to have to record that way. And, paradoxically, that's why you'd have flawed pitch, sometimes, out-of-sync, so on - they did it all in full takes. Not piece by piece, like today, word by word, etc., but just going through it. It got more fragmented going towards the 80s, technology allowed for easier comping, but still. You had to have great chops to do that, even if you occasionally f... up, it's inevitable, but they pulled it off. And when you had somebody with flawless performance, like Coverdale, or Sting, that was the mark of true mastery.

1 hour ago, Teegarden said:

And just like you say about the every day consumer products, in the analog studies there's been tons of development to get sounds perfected.

Precisely. Not just analog gear - although the very act of miking things up, and running through circuitry had a beautiful effect right of the box - vibrance, variance, compression, saturation, hi-end roll off for tape - but beautifully crafted analog gear. But you need to take time to listen without bias, and not attach your identity to some words. Not have a nervous breakdown if someone calls you "unmodern" or anything of the likes.

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