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Posts posted by Bruno de Souza Lino
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As it's customary with Intel 10th gen cpus, check your thermals. These SKUs are known to run quite hot and enter throttling very often. If that's the case, consider undervolting them.
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7 hours ago, John Vere said:
I’ll add to this that while it’s best practices to install in order of release version I just went through a process of digging up missing components for my second computer I hadn’t been using much.
A year ago I rebuilt my office machine and only installed CbB and the basics that came with it.
A bit later I had to run CCC and installed everything from Splat.
I found myself wanting to some editing on this machine and as the weeks went by I kept finding missing plug ins. Most are simply a matter of grabbing the installers but some were tied to old versions.
i had no problem installing as far back as 8.5 and getting stuff like v vocal. So bottom line is I don’t think it is the end of the world if you don’t go in sequence and find later on needing something else. Just pay real close attention to the dialogue as you install the old version.
V-Vocal can be found in up to X2, it being the last Roland version of SONAR.
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Even though you see purchase links and prices on the Cakewalk website, clicking on any of them will lead you to this page:
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15 minutes ago, pcooke9 said:
Even when you set the First and Other Beats to the same sound and number in the UI, the first beat is still a few dB louder. (~5-6 dB louder)
In that case, add a brickwall limiter to the metronome bus.
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You might also want to install X3 to have Perfect Space, Vintage Channel and some of the synths that were left behind, like Pentagon.
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The last version of SONAR to have V-Vocal is X2. After that, Cakewalk was sold to Gibson and all the proprietary Roland stuff had to go. V-Vocal was replaced with Melodyne in X3 and remained like that until Platinum.
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Set first and last beat to the same sound and same volume.
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18 hours ago, azslow3 said:
I guess in case that is true, my old 8x8 USB2 interface couldn't work on it's lowest settings when connected to 10m USB hub in parallel with several USB1 devices. But it works. USB specification deals with different standard/speed devices much better then making everything slow. ?
Also under 1ms RTL is never "comfortable". Computer should be top optimized and plug-ins carefully selected. Yes, there are no USB interfaces with such feature. But 3.3ms is really usable, with USB2 and moderate buffer size. In practice, the difference can be rarely perceived (taking into account that moving your head 30cm in any direction change latency by 1ms...).
That's because some PCIe lanes speak directly to the CPU by means of CPU interrupts. The same can be talked about PS/2 vs USB.
17 hours ago, John Vere said:People and possibly the manufactures are obsessed with RTL specs. What does it matter if you track using direct monitoring? I think stability is way more important to me.
Lower RTL is only needed if you want to use things like Guitar SIMS playing live in real time.
My system had a reported RTL of 27 or so for most of the last 12 years. It's only recently down to 9ms. I have always kept my buffers at 256 and everything always sounds great. And because I have always used direct monitoring I experience bang on timing as I play.
Funny story about the real world latency situation. As a bass player I always felt I played tighter with the drummer if I stood right there next to the kit. I'd often have one leg right in front of the kick drum. When I first started recording ( analog days) I found I could not get that bass to lock in to the timing unless I put the studio monitor next to my ear or used headphones. So to me that 5ms delay when you are ? feet from the speakers/ monitors counts.
For live music one huge benefit for modern performers using In ear monitors and not much talked about, is no more latency on stage.
Sure. Any latency below 5 ms is considered real time by human ears. In reality, we have much more latency that that. Humans can cope quite comfortably with up to 40 ms of latency depending on scenario and you get 1ms of latency for every foot your ears are away from the sound source. If you're playing 10 feet away from your bass amp, you're already dealing with 10 ms of latency.
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Cakewalk by default records at 32 bit, which has about 150 dB of dynamic range. Windows is set to 16 bits by default, which has 96 dB of dynamic range. There is a chance that your soundcard has some sort of gain compensation feature that's doing that to prevent the export from clipping.
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On 1/2/2021 at 6:17 PM, John Vere said:
My observation is that generally a well behaved DAW will go hand in hand with the Quality of the ASIO driver supplied by manufacturers of interfaces.
10 year ago USB ASIO drivers seemed to suck if you had a lower price interface.
And PC specs used to also matter. Notice that there’s no point asking “ What are your computer specs?” anymore when troubleshooting someone’s issues. It will be rare to find someone with an under powered machine these days. And generally you can get away with breaking the rules we had 10 years ago like shut down the internet as it seems to make little difference with modern systems
Seems most mainstream audio interfaces have good drivers now. And I’m impressed by company’s like Focusrite who upgrade drivers for their older units. Seems very few people have issues anymore and when we do they are that much harder to pin down
Expecting everyone to have current or similar hardware is a dangerous assumption to make. One example would be Spleeter. It makes use of TensorFlow to work, but their initial build assumed everyone using the software had nVidia graphic cards, and TensorFlow was compiled with CUDA support enabled. Many people with AMD cards couldn't use the product unless they compiled TensorFlow from source without CUDA enabled. After that was fixed, many people started complaining that Spleeter wouldn't work and there were a whole bunch of errors. The developer didn't say anywhere that Spleeter doesn't work if your CPU doesn't have AVX instructions. The Appleseed Blender render did a similar thing with SSE4.1 but it just crashed Blender during rendering instead of throwing an error.
There's also the argument of maximum performance. Not having things like internet on while you work or disabling some services could be the difference between being able to run one extra instance of that plugin or having to increase your latency samples. Windows will use all the resources from your machine it can to do its tasks without your permission and that's not the lack of control over my hardware I wish to have.
As better as ASIO drivers are nowadays, USB still is a serial bus. If you have a device that's slower than your interface on the same bus, the controller will run everything at the speed of the slowest device in the bus and there's nothing you can do about it except for making sure your interface has its own bus and nothing else uses it. I'm yet to see a PC + USB interface combo that can match a PC + RME Hammerfall HDSPe combo. With the latter and a sufficiently fast PC, you can quite comfortably run under 1 ms of latency with little to no performace hit.
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AFAIK only ESi had ASIO drivers that could make use of multiple interfaces. Another option is ASIO4ALL.
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That would solve one of the issues with CbB's routing, which is the hard pairing of outputs. Every single output is paired 1,2, stereo 1 and 2, 3,4 stereo 3 and 4 and so on. That's all fine and dandy until you find a plugin which has, say, outputs 4 and 5 as a stereo pair. With the current layout, it's impossible to create a single stereo track that only contains these two routed left and right.
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Presonus has a very good guide on tweaking your Windows PC for audio. There should also be a Creative Sauce video on that.
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I would wait to switch. AT5 is slower and has some sharp edges still.
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That would also boil with the fact that people make music with their ears and not eyes so if it sounds good, in phase or not, it's what matters.
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On 11/25/2020 at 4:24 PM, abacab said:
Z3TA+2 and DimPro have individual installers in Command Center. I think for the original classic z3ta+ that was bundled with certain versions of Sonar, you would need to run the Sonar installer and then select that plugin from the bundle.
scook is correct, none of them are locked to Sonar.
For Platinum, the installation is automated and you have no control over what gets installed other than knowing which gets installed in each of the 10+ packages in Command Center.
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Would be possible to provide a download source that's not MEGA?
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On 12/29/2020 at 7:31 AM, cclarry said:
Worst purchase...anything IK.
I love their plugins, but everything they do business
wise is terrible, not to mention that IK plugs have almost
ZERO resale value...and that doesn't include the Jampoints
they take BACK when you sell one of their products...
Their business model is making them one of the WORST investments
a person can make...with Waves coming in a close second...You are forgetting the 180 day expiration date for sound downloads. Expired? You have to pay 10 bucks to have the download again for 180 more days. Not only that's dodgy as heck, but you can also have sounds for free products expire as well.
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42 minutes ago, pwalpwal said:
lite version of lab? i thought lab was already a lite version of the synths?
Not really. Lab works like a workstation pretty much. If you have the full version of the synts used, you can edit the synth parameters and also build presets using multiple synths and such. Think of it like Kontakt or Mainstage.
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As a side note, if you happen to save your track setup as a track preset with a drum map to go with it, that drum map is not loaded when you load your track preset. It just adds a new empty DM entry that you have to reload your preset.
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- Ability to replace FX Rack plugins like you can to with Virtual Instruments and ProChannel modules.
- Either the ability of loading ProChannel as a sort of Channel Strip plugin with the same functionality of the standard or load ProChannel modules as plugins.
- Allow us to hide some, if not all, track meters.
- Ability to bounce audio coming into aux tracks without having to manually record it.
- Move some of the settings to more localized places instead of having everything crammed up into the Preferences menu.
- A more flexible and modern version of ACT. The current version is complicated for the wrong reasons, unintuitive and clumsy to set up and use. If you want a good example, look at how Studio One does it.
- A more flexible and modern version of Drum Map Manager. The current version is complicated, unintuitive and clumsy to set up and use.
- Improvements to the PRV with things such as ability to do pattern selection (selecting every other note, etc.), proportional editing and so on.
- More flexbile audio routing.
- LFO an an option for automation envelopes.
- Some sort of Listen Bus or Monitoring track.
- Improvements on the Metronome.
- Improvements on latency handling.-
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If you're using Amp Simulation, it's a better idea to gain stage at the plugin's master volume, as hitting the plugin at a lower volume will change the sound that comes out.
Also, your clipping might not be caused by volume, but frequency buildup. Grab a good spectrum analyzer, like Voxengo SPAN and check it. Or use Dan Worral's method:
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I've disabled auto-save since it crashed the DAW the first time it happened. That is a bit more strain on you, as you have to manually save, but no more issues and I can even save with the project running.
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Another alternative to consider would be ValhallaDSP Supermassive and Dragonfly Reverb, both free.
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Audio Flash Drives?
in The Coffee House
Posted
The quality differences are not identifiable by humans ears unless you measure the audio and know which is which, but that opens the door to perception bias, confirmation bias and other cognitive issues.
It's hard to suggest that people didn't think about audio quality when they developed the MP3 format 28 years ago.