Bruno de Souza Lino

My mistake. Vintage Channel is on X2. X3 is for Perfect Space, LP 64 EQ and MB as well as Pentagon I.

Oh yes. There are quality differences between speakers, but that's not what I'm talking about. I'm talking about the notion that all it takes to tell MP3 from WAV is a sufficiently high end playback system. Should I remind you that nobody gave two tosses about Yamaha NS10's until Quincy Jones made a Platinum record using them? Gear quality is not solely tied to how much it costs. You'd be surprised to find boutique hifi speaker systems costing over half a million dollars with up to 25% THD and IM Distortion. That what believing in audio gets you.

Once again, if their ears are so special, why their detection relies on a specific set of conditions to work? As per mixing on 10$ headphones, I wouldn't be surprised to find many people out there doing it. Not everyone has the budget to high end equipment but have a will to make music. Mixing with headphones was almost taboo 10 years ago and now even famous engineers are doing it.

- Better handling of clip fx. The way it's currently set up, you have this teeny tiny rack that lists them and you can easily lose it by clicking somewhere else.

The latest version of Essential removed the Audio to MIDI conversion feature. You need the next version high up for that.

I'd only use the vouchers if you could use them to purchase stuff that's the same price as the voucher itself or less than it.

If the difference depends on a playback system, then I'll argue that knowing the (alleged) quality of the playback system introduces confirmation bias. If those special eared people can detect it, they can detect it no matter what the conditions are. That would be the same as someone with perfect pitch stating they can only accurately tell pitches under certain conditions.

I managed to download the piano plugin and could not get a clean sound out of it. All sounds come out distorted and full of noise no matter how high the samples were. On top of that: - Your website states that the plugins are VST3. They're not. Putting them where VST3 plugins are and scanning them yields no results. - Initially I thought you had to download the plugin and the samples separately, then I was surprised to find the plugin along with the samples in the samples download. What? Not only that, but the dll bundled with the samples has a different name and is detected as a different plugin too. - There are no installation instructions for the samples anywhere. As for the UI: - The UI is very dark and has contrast issues. People that are visually impaired will have a hard time using the plugin. - Many of the buttons are unresponsive when you click on them. When I opened the settings to see if I could fix the audio issues, the plugin would not close it no matter how much I clicked on the button. - The presets section seems to have a cork, as clicking it the first time doesn't load a preset, but rather shows "ScriptLabel1" instead.

Without knowing the exact parameters of your test environment, I highly doubt it. Ethan Winer has some tests on his website for bit rate, dither and so going on for over 10 years and no one has been able to guess the results correctly.

This song was originally meant to be a submission for a song contest hosted by Orchestral Tools with the theme "outside", but I missed the deadline to send it. I don't quite remember what I used for guitar, but the drums are DrumGizmo making use of its humanizing feature with some manual tweaking of velocities and timing, as well as adding things like ghost notes and such. Spectrogram was done with ffmpeg using some pretty large commands that spit out the video with the audio added to it and the effect.

The song sounds nice and has a nice atmosphere to it, but it's a bit too flat IMO. What I mean by that is the volume remains constant throughout the piece even in places where you should be louder or softer. You should consider that rather than just making everything louder. I don't think it's the kind of music that would benefit from having everything with no dynamic range, considering the style. As per the drums, they do sound sequenced. A real drummer, no matter what their name is, will never hit the exact same volume every time. Even when you compress the sound you can hear that. There's also the timing thing. It's impossible for a human being to perfectly lock into a time grid and even some of the drummers lauded for their tight timing will never hit perfectly on time. Granted that my impression might stem from the mix being static and not pushing and pulling. If you're programming drums by hand, getting them to sound realistic manually is some extra work but can be worth it, rather than expecting a plugin to get it right.

I had to complain about it in their forums about their eLicenser expiring my Wavelab license. Eventually, one of the admins gave me a new activation key. The only thing I'd consider if I had extra money would be the 50 EUR upgrade from LE 10.5 to Elements 11.

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.

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.

V-Vocal can be found in up to X2, it being the last Roland version of SONAR.

Even though you see purchase links and prices on the Cakewalk website, clicking on any of them will lead you to this page: http://www.cakewalk.com/announcement

In that case, add a brickwall limiter to the metronome bus.

You might also want to install X3 to have Perfect Space, Vintage Channel and some of the synths that were left behind, like Pentagon.

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.

Set first and last beat to the same sound and same volume.

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. 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.

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.

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.

AFAIK only ESi had ASIO drivers that could make use of multiple interfaces. Another option is ASIO4ALL.

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.