Glenn Stanton

it's not so much complicated as it is preparation work so your projects are in the best condition possible when sharing them, otherwise the recipient has to do all the work, and they could get it wrong resulting in a lot of wasted time. plus, you're likely to do a bunch of these steps when it comes time to archive a project. knowing if you rehydrate it, you'll mostly have things organized in a way that helps you get back to mixing instead of patching up a mess...

hmmmm. i use BT speakers and sometimes are headset with CbB many times over the past couple of years. if you have them installed as Windows devices such as audio outputs, recording sources, etc then they (generally speaking) can be seen in CbB using WASAPI or ASIO4ALL (in this latter one i find it annoying that it sometimes reverts to the Bose headset when i had set it to the USB audio, but its easily changed in the admin console).

congrats on the new specialized high intensity training (S.H.I.T.) job! ? i tried this about 8-9 years ago with a fairly large server-size system (HP DL w/ 160 cores on 8 CPU , 2TB RAM, etc) and oddly enough it had a built-in audio card. running Windows Server 2012. so since the $50K machine was sidelined from a project, i loaded up a VM using SONAR PE (X2) and while i could not "hear" via audio card, since it was present in the VM, i could hear and mix via remote desktop (via 1gb hardwired network) from my laptop. BUT (always a big but somewhere) it was flakey and had the crackles etc you'd expect on a low end PC, and even with a bunch of tweaking i could not get it to be happy. maybe the Windows 8 on a VM was not happy. whereas my Dell 545 (still running today - 3 years ago updated w/ quad core cpu , ram, and OS) was perfectly fine with the same software, OS, and general settings. maybe with more modern OS and Hyper-V updates it will work.

if you're planning on sharing - some preparations are needed to do this smoothly: make sure your audio clips are bounced to clip and extended (not dragged!) to 1:0:0 mark so the person receiving them can align them to the zero mark. yes, you can export them individually as broadcast wav which have the time markers embedded, but what if the recipient cannot use that? so zero mark is best. if you know the person on the receiving end will not have the plugins needed, print a separate track with the effects and then remove them from the project once you have documented their settings. make sure any plugins which are yours exclusively and not likely to be available (easily) for the other person (and which you determined are critical to the mix) are documented. Word is a good choice with screenshots. make sure all the clips and associated file names are understandable. it makes it much harder for someone to sort through clip#99 clip#A mix_clip#5 etc create a copy of the project with it's own audio file directory. this (check the copy audio files box) makes copies of the audio clips and only the ones you're actually planning to share and not a variety of clips which will be confusing to someone needing to patch them together. this is where you can export OMF file as well - open the newly saved-as project and export. ensure the person you're sharing with can read your files at the sample rate and bit depth of your project. it won't be helpful if the person getting your files can only support 44.1K @ 24 bit to send files which are 96K @ 32 bit... if not, convert the audio files to the required sample rate and bit depth. export a "draft" mix (or demo/reference etc mix) so the person receiving the package have a rough idea of what you're doing at the time you sent it... then zip the new project directory. this captures the project file, OMF, Word doc, rough mix, and audio clips into a single blob which is universally accessible. the recipient of the package can use the project file or just import the audio clips into the DAW of their choice. now share the blob via USB drive, or via cloud services. of course, if the person you're exchanging with use the same version of CbB, then you can skip the OMF and leave any shared plugins in place. feel free to have people sending you their projects take the same efforts to package up their work before sending it. ?

sorry for your loss. it's actually only 5 tracks - drums, bass, guitar, keyboard, and vocal. so no doubled parts, maybe it's the constantly shifting tempo and some flanger (guitar) and chorus (bass) causing the pitchy bits (not discounting my singing of course!) yeah, it is a bit sloppy in parts. mostly because i started out testing the tempo mapping by dragging the guitar (played without click or other parts) onto the tempo bar, then adding a midi drum pattern and playing the other bits.

so why would you have to drop CbB if you buy into PT? i'd go out on a limb and suggest many people on this forum (and many more not on the forum) have at least 2 DAW they use, and probably many have 3 or 4 DAW products (and likely all kinds of audio processing tools as well). for myself (having 6 DAW products), if i get audio files and no restrictions, it's CbB. if it's a mostly mixed PT project, then i open it in PT, save in PT, and re-share it back as a PT project. same for Cubase, FL, REAPER, Audacity, Studio One, etc etc... it would be silly to use a framing hammer to assemble picture frames when an air powered brad gun would be perfect... the right tool for the job is the right tool. smae for plugins - i need both VST and AAX so investing in products which support both eliminates redundancy in the plugins i use and the breadth of knowledge needed to use them across DAW... if you're making a living off this stuff, then investing in the right tools and acquiring the skills to use them is part of the gig...

i'm guessing but the Park Control app you're using is made by the same people who make Lasso ? Lasso is so much more. however, it's mainly for technically inclined folks who want to get deep under the covers of CPU affinity and many other bits. yes, if the CPU is kept full on, full power, it will be more heat than allowing the OS to manage the CPU utilization.

yep same here. plus with RX7 (or later, perhaps earlier as well) - de-reverb, de-noise, etc all help. on the simpler "synthetic" stems - capture a MIDI of each from Melodyne, use it for soft-synth replacements, and just keep the vocal or lead solo ? (sort of like using new paint on a Sistine Chapel ceiling when needed ? )

+1 on the Gibraltar thrones. i have two round ones. way way better than "saving money" on a cheap one. i use one for keyboard and one for drums. very heavy but much more comfortable if you're in the seat for a couple of hours.

new outro theme for dogs gone to heaven movie updated version ->

if you look into the "standards" around this, there are many partial attempts over the years and 4-5 which seem to have persisted. it's likely that some software companies probably invested in any number of the partial ones as well as the current "standards" (specifications really) and are also the ones likely to be involved in the committees to create the standards (both for showing their love of the movie industry, and of course to lock out competitors). so it's not surprising that the implementations vary enough to prevent easy re-use across competing products whilst each company is hoping they become the favorite... somewhat interesting is the choice of Microsoft file format (via COM interface) for the storage blobs (alleging this works seamlessly across Mac and Linux... fully possibly given COM is simply a bit-wise interface specification re-usable binaries as long as endian direction is taken into account ? )

i know for my licensed version, when i download it, it's customized install package w/ my license built into the installer data. no sure if the demo software works in a similar fashion, if so, the "date" might be embedded as part of the installer package so regardless of when you install it, it might expire in 30 days from the time you download. maybe reach out to Celemony and see how it works and perhaps get a refreshed demo.

is AAF primarily used because of dialog timing being preserved while the video is edited and sync'd to the external audio processor? or does it have a place for the background music sync'ing as well? which to me would be less useful as most soundtrack composers tend to write/perform with the latest and greatest major edits so the music transitions are closely matched to the onscreen activities and director cues - versus being cut up and likely discontinuous a a result whereas dialog would be "tight" to the acting and less impacted across edits... my 2¢ as i don't do a lot of video on-the-fly composing/performing.

Lasso has a lot more configuration options for setting up application specific bits so you can set core distribution, latency of applying restrictions, and ignoring total cpu for restriction, etc. all of which can help avoid having CPU do slowdowns or re-schedule for intermittent or short term processing spikes etc

do a reinstall. something got dropped. make sure you're using the latest and greatest installer 2021.01

i would go with the park control until you update to W10. and even then i'd get the Lasso tool ?

some good info on wide guitars: https://www.puremix.net/blog/double-tracking-rock-electric-guitars.html

in general, unless the source is stereo (e.g. keyboard direct in, drum overheads or room, guitar stompbox chorus effect, etc) record it in mono. on outboard effects applied to an instrument, it's always a good idea to record another track clean in case you want to "re-amp" later or just overlay a clean version etc. for MIDI capable instruments, record the MIDI as well. you don't want a good performance to not be captured with all options for using it in the mix. you can always delete unneeded stuff later if space or organization is a concern.

in general, if you have discreet mono tracks, LCR is just that. to widen a given instrument in stereo you can use mid-side EQ, delays (Haas effect) or have it recorded with a stereo mic setup (recommend M/S setup so it can collapse in mono). listening to the mix in mono (meaning tracks, busses, and a single center speaker all mono) allows you to find places for everything using panning and volume and EQ. then opening it all back up to stereo is often an amazing thing ? however rechecking in mono is important. why? most night/dance clubs use mono on their systems to avoid weirds holes in the music or other audio, people often share one side of their ear buds etc. so if the balance is good in mono or fairly even in ear side, your audience will often benefit from that effort. the Ozone Imager uses M/S EQ and has a stereoize Haas effect. which can result in phase issues. poorly done mic'ing can result in phase issues and time issues. skewing a double track (even if played fairly closely) can lead to what sounds like phase issues since it mimics a poor mic setup. and spanning an instrument via a delay can result in a "big mono" sound and may be what you want to not. so setting a stereo instrument within a stereo track (e.g. a leslie organ) means you need to have a means of controlling each side of that track with a panner or a plugin (like channel tool) to apportion it within the sound field (e.g. the leslie spans C+10% to L+60%)

if the tracks and busses are all set to mono and things are still "wide" then i'd look at: - windows sound settings - disable all enhancements; - speaker placement and make sure all DSP is turned off. having "room correction" running when the room is badly in need of fixing is going to result in poor stereo monitoring. as a general rule all mono input to the speakers should result in a stable center phantom image. if not, then something is wrong in the speaker path, placement, or room interference causing a tiny sweet spot (meaning even small head movements are catching phase shift effects) as a result of comb filtering etc. once that is working correctly, switching the DAW back to stereo and finding the plugins which are causing the "widening" effect should be much easier.

and then it's all crushed down into a 128K MP3 to be listened to on tiny ear buds ?

yeah, i have a bunch of old synth plugins which are 32-bit and i like to keep around. as more 64-bit synths come out that are better sounding i'll make the switch. so far JBridge is handling things perfectly for me.

JBridge, and then don't worry about 32-bit. worth every $. https://jstuff.wordpress.com/jbridge/

if the mixer has a USB IO connection, it can generally be used for your IO. if not, you could take the main output of the mixer into the 2i4 IO, and the outputs of that into an aux return on mixer or into a pair of line level channels (caring not to set up the routing to cause feedback loops). most times a mixer has only stereo in/out on the USB unless you buy a high end unit which supports multiple channels. as an example - i have an old Behringer MX2004A which has inserts on the 8 mic channels, so i normalize via patch panel the IO of the UMC1820 from the mixer which has better preamp levels for mics although when i want something really clean, or i am using a line out keyboard etc,, i patch in directly to the UMC inputs. since i'm only using the mixer for inputs, i route the outputs from the UMC into my cross-over and then monitor switcher which feed the powered monitors, the amp for the 901s, or the optical out to the 5.1 receiver.

if the USB mic is the source in CbB, then unless the mic has it's own builtin volume control, the raw input into the DAW will be 100%, so "turning it down" in your monitoring doesn't solve that problem. best bet - get yourself a 2 input IO unit with XLR and an SM57 or SM58. probably total under $200. as far as vocal-room interactions - you could build a simple set of frames - tied together - and hang heavy packing blankets of them. if the frames are tall enough, hang a packing blanket over the top. leave some gaps for air. some ideas: https://3dwarehouse.sketchup.com/model/d7c034208826744bfda112de017f7f93/Recording-Studio-Inexpensive-Vocal-Booth https://3dwarehouse.sketchup.com/model/1ed68da16d67650b6d2f7f4796305f20/Cheap-Vocal-Booth-Redux