OutrageProductions

My CPU/MOBO is only 18 months old, my SSD's are about 2 years old (2Tb each - times 4), so it's not an old computer by any means. I'm running an i9k processor at 3.4Ghz, 64Gb ram, 32Gb GPU ram, two GPU cards, quad video monitors. I update the CPU and MOBO about every 3 years. M-Audio actually discontinued the CS support software at the end of XP, and it wouldn't load at all from Vista onwards. I have all functions of the channel strip; encoders, SEL, REC, SOLO, MUTE, and moving touch faders. The CS software (under XP) had the ability to transform the surface into a full DAW controller - plugins, editing, etc. Frankly, I don't miss that part too much. But the transport controls, scrub, window zoom functions, locate to markers, set and arm loops, etc. are all very handy at my left hand while I track-ball with my right hand. There was some tweaking that I had to do to the standard Mackie Control Surface API tho. Can't help you with the 44.1 sample rate thingy. I haven't done anything other than 48k/24b (most of my job is scoring to film/video) for over 20 years. The ProjectMIx was a very unique animal, and I love it. However... being in use 24/7/365, I'm on my third power supply since I bought it in 2003.

1. YES, 5.10.0.5061 (all of my crashes are due to the fact that I know the software so freakin' well that I can operate it faster than it can respond, and so I occasionally outrun the keyboard input buffer. When it crashes that way (maybe twice a month, and I work on it 6 hours everyday) I just take my own advice and go get a fresh cupper... 2. I have an old Belkin 3 output 1394 TI chipset card running the MS driver 10.0.19041.1 (class compliant) 3. Yes, tho I have most of them muted (startup active, network ID active, mouse cliks/errors/et al muted) 4. I run all of them through 3/4 most of the time which is my headphone out from PM, also have Nearfield mons out of 1/2 just by swapping mutes in CbB. Can also fairly easily switch to my 5.1 Surround rig in CbB by switching the HW output mutes to my ADAT outputs. 5. EVERYTHING is running at 48k/24b [Windows, YT, Browsers, VB Audio Voicemeeter (for Skype/OBS/Zoom, etc.] Here I have the faders linked and group muted. I switch from headphones to nearfields to Surround mains just by Mute/UnMute. Levels are controlled by and external surround monitor controller that I designed and built. I also have my TV running in to ANA 7/8 and directly out to 1/2 (nearfields), but that is just for background noise when I'm editing. I have the transport functions available on the PM, but, alas, they eliminated the software and driver for the entire MIDI control surface functions when Vista when bye-bye. Sad. But the faders/rotary/monitor levels work, and everything from Locate and below is functioning. But the Encoder Operations section, which was handy when a complete MIDI CS are just pretty to look at now...

Not directly in the DAW, however... if your mixer has MIDI out or control voltage out, you could see if it can be programmed to send a signal to the DAW similar to a footswitch. I've used a Normally Open footswitch to 'punch in' record, but it has been many years ago.

I'll chime back in again on this; As in previous post on this thread, I have ProjectMix working (still!) on Win10 with Win7 drivers in compatibility mode. ASIO all the way. RT latency at 10.4 milliseconds. I have Win system sounds, Skype, Zoom, OBC, Chrome, YouToob, Cakewalk, Nuendo, Cubase, and my video edit sled all going through it with no issues. I select the hardware outputs in the Windows Sound module. Been doin' it this way since 2005.

This is incorrect. Let's dispell a few common misconceptions; The velocity of a sound wave is 343 meters per second at sea level, regardless of frequency. Other environmental factors will affect it, like temperature, relative humidity, and barometric pressure. However, what you experience as 'behind the groove', when at a distance from the source (drummer) is a time delay caused by the transmission line (air). Musicians that play with headphones or IEM's live, will play 'in the groove' at any distance relative to one who is listening to the direct and/or reflected source. Fun fact: most audiologists and acousticians agree that approximately 33 milliseconds is the average human threshold for a discernable "first reflection", which is why, if your hardware & DAW are capable of running with a round-trip latency of less than that, you will be able to play relatively 'in time' with the groove when recording, especially if using headphones. Relating to the OP post; loudspeaker directivity is narrower in higher frequencies than low frequencies, which means the lows have more opportunity to reflect the waveform off of surfaces during the transmission (out of the room and down the hallway, as it were) causing multiple delay effects and reflections to be perceived, leading to a 'smear' of the timing.

What is the purpose? 1/4 of a 60 hertz signal of any waveform is 15 hertz, which will only be audible by whales and pachyderms, and 1/8 of a 60 hz signal is 7.5 hertz, which can generally only be detected by a seismograph, even if you had an extreme low frequency electro-mechanical transducer.

This is totally the result of the acoustical response of the room/environment that you are listening in. I've spent 30 years as an acoustical engineering consultant, and the only thing you can do is construction and treatment, which is not inexpensive. Learn to listen to stuff that you know as a reference, then compare. You'll notice lots of things that are anomolies. Live with 'em. Or build yourself a real, treated, monitoring environment.

You might think about getting Audacity instead. Much easier to learn.

There are several more modules that can be applied to show up in the top toolbar. You have yours justified currently. RMB in any blank space in the toolbar and select what you want, then save as custom workspace.

Most of the collaborators that I work with around the world are using DistroKid, and seem to be very satisfied.

Planned obsolescence is the modern way. Studio One, Reaper, etc. kicking & screaming.

Reading is hard, Comprehension even more so.

I have several project templates built with Scaler on track 1 driving other instruments. Make sure that the instance of Scaler is inserted as a simple instrument and that you have initiated the VSTi MIDI output from it, then it will show up in the MIDI input list of other instrument tracks or MIDI tracks. It gets a little trickier if you want to use the Multi-out of Scaler, but the flow is the same.

My inventory tool was old, I only ran it once a couple years ago, and I don't have more than 148 plugs, 60ish of which I never use anyway, so I couldn't care less.

It will only find VST2 versions (DLL's), not VST3.

When editing in the PRV, if I select copy & paste or duplicate, my Articulation track gets copied with the notes clip and Automation lanes. You must have a setting that is incorrect.

Here's one from far out in left field...

Bye Felicia!

I had that exact MOBO until 2018. It performed well under WIN7, but started to groan under Win10 so I upgraded the MOBO & CPU. It should still be serviceable, but you won't win any races with it. ?

Check the cleanliness & integrity of the video connection to the monitor (be it DVI/HDMI/USB/other) for a good contact. Sounds like a possible hardware cable issue.

Recording with the internal audio device will always be worse with regard to latency. Save up for a small external Audio I/O interface with genuine ASIO drivers. You'll never look back. I've used one of these for years on remote work and it's been a trooper.

95% of my current work is locked to video, and I concur with @Andres Medina on the format. I can lock to 4k HD video in CbB with nary an issue with my playback on a 27" screen #4. I've never run across your LAV decoder...

I concur on this.

The NI walk-through on YT covers everything you need to know.